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-rw-r--r--src/modules/echo-cancel/adrian-aec.c275
-rw-r--r--src/modules/echo-cancel/adrian-aec.h382
-rw-r--r--src/modules/echo-cancel/adrian-aec.orc8
-rw-r--r--src/modules/echo-cancel/adrian-license.txt17
-rw-r--r--src/modules/echo-cancel/adrian.c117
-rw-r--r--src/modules/echo-cancel/adrian.h31
-rw-r--r--src/modules/echo-cancel/echo-cancel.h90
-rw-r--r--src/modules/echo-cancel/module-echo-cancel.c1778
-rw-r--r--src/modules/echo-cancel/speex.c115
9 files changed, 2813 insertions, 0 deletions
diff --git a/src/modules/echo-cancel/adrian-aec.c b/src/modules/echo-cancel/adrian-aec.c
new file mode 100644
index 00000000..e969e8c5
--- /dev/null
+++ b/src/modules/echo-cancel/adrian-aec.c
@@ -0,0 +1,275 @@
+/* aec.cpp
+ *
+ * Copyright (C) DFS Deutsche Flugsicherung (2004, 2005).
+ * All Rights Reserved.
+ *
+ * Acoustic Echo Cancellation NLMS-pw algorithm
+ *
+ * Version 0.3 filter created with www.dsptutor.freeuk.com
+ * Version 0.3.1 Allow change of stability parameter delta
+ * Version 0.4 Leaky Normalized LMS - pre whitening algorithm
+ */
+
+#include <math.h>
+#include <string.h>
+#include <stdint.h>
+
+#include <pulse/xmalloc.h>
+
+#include "adrian-aec.h"
+
+#ifndef DISABLE_ORC
+#include "adrian-aec-orc-gen.h"
+#endif
+
+#ifdef __SSE__
+#include <xmmintrin.h>
+#endif
+
+/* Vector Dot Product */
+static REAL dotp(REAL a[], REAL b[])
+{
+ REAL sum0 = 0.0, sum1 = 0.0;
+ int j;
+
+ for (j = 0; j < NLMS_LEN; j += 2) {
+ // optimize: partial loop unrolling
+ sum0 += a[j] * b[j];
+ sum1 += a[j + 1] * b[j + 1];
+ }
+ return sum0 + sum1;
+}
+
+static REAL dotp_sse(REAL a[], REAL b[])
+{
+#ifdef __SSE__
+ /* This is taken from speex's inner product implementation */
+ int j;
+ REAL sum;
+ __m128 acc = _mm_setzero_ps();
+
+ for (j=0;j<NLMS_LEN;j+=8)
+ {
+ acc = _mm_add_ps(acc, _mm_mul_ps(_mm_load_ps(a+j), _mm_loadu_ps(b+j)));
+ acc = _mm_add_ps(acc, _mm_mul_ps(_mm_load_ps(a+j+4), _mm_loadu_ps(b+j+4)));
+ }
+ acc = _mm_add_ps(acc, _mm_movehl_ps(acc, acc));
+ acc = _mm_add_ss(acc, _mm_shuffle_ps(acc, acc, 0x55));
+ _mm_store_ss(&sum, acc);
+
+ return sum;
+#else
+ return dotp(a, b);
+#endif
+}
+
+
+AEC* AEC_init(int RATE, int have_vector)
+{
+ AEC *a = pa_xnew(AEC, 1);
+ a->hangover = 0;
+ memset(a->x, 0, sizeof(a->x));
+ memset(a->xf, 0, sizeof(a->xf));
+ memset(a->w_arr, 0, sizeof(a->w_arr));
+ a->j = NLMS_EXT;
+ a->delta = 0.0f;
+ AEC_setambient(a, NoiseFloor);
+ a->dfast = a->dslow = M75dB_PCM;
+ a->xfast = a->xslow = M80dB_PCM;
+ a->gain = 1.0f;
+ a->Fx = IIR1_init(2000.0f/RATE);
+ a->Fe = IIR1_init(2000.0f/RATE);
+ a->cutoff = FIR_HP_300Hz_init();
+ a->acMic = IIR_HP_init();
+ a->acSpk = IIR_HP_init();
+
+ a->aes_y2 = M0dB;
+
+ a->fdwdisplay = -1;
+ a->dumpcnt = 0;
+ memset(a->ws, 0, sizeof(a->ws));
+
+ if (have_vector) {
+ /* Get a 16-byte aligned location */
+ a->w = (REAL *) (((uintptr_t) a->w_arr) + (((uintptr_t) a->w_arr) % 16));
+ a->dotp = dotp_sse;
+ } else {
+ /* We don't care about alignment, just use the array as-is */
+ a->w = a->w_arr;
+ a->dotp = dotp;
+ }
+
+ return a;
+}
+
+// Adrian soft decision DTD
+// (Dual Average Near-End to Far-End signal Ratio DTD)
+// This algorithm uses exponential smoothing with differnt
+// ageing parameters to get fast and slow near-end and far-end
+// signal averages. The ratio of NFRs term
+// (dfast / xfast) / (dslow / xslow) is used to compute the stepsize
+// A ratio value of 2.5 is mapped to stepsize 0, a ratio of 0 is
+// mapped to 1.0 with a limited linear function.
+static float AEC_dtd(AEC *a, REAL d, REAL x)
+{
+ float ratio, stepsize;
+
+ // fast near-end and far-end average
+ a->dfast += ALPHAFAST * (fabsf(d) - a->dfast);
+ a->xfast += ALPHAFAST * (fabsf(x) - a->xfast);
+
+ // slow near-end and far-end average
+ a->dslow += ALPHASLOW * (fabsf(d) - a->dslow);
+ a->xslow += ALPHASLOW * (fabsf(x) - a->xslow);
+
+ if (a->xfast < M70dB_PCM) {
+ return 0.0; // no Spk signal
+ }
+
+ if (a->dfast < M70dB_PCM) {
+ return 0.0; // no Mic signal
+ }
+
+ // ratio of NFRs
+ ratio = (a->dfast * a->xslow) / (a->dslow * a->xfast);
+
+ // Linear interpolation with clamping at the limits
+ if (ratio < STEPX1)
+ stepsize = STEPY1;
+ else if (ratio > STEPX2)
+ stepsize = STEPY2;
+ else
+ stepsize = STEPY1 + (STEPY2 - STEPY1) * (ratio - STEPX1) / (STEPX2 - STEPX1);
+
+ return stepsize;
+}
+
+
+static void AEC_leaky(AEC *a)
+// The xfast signal is used to charge the hangover timer to Thold.
+// When hangover expires (no Spk signal for some time) the vector w
+// is erased. This is my implementation of Leaky NLMS.
+{
+ if (a->xfast >= M70dB_PCM) {
+ // vector w is valid for hangover Thold time
+ a->hangover = Thold;
+ } else {
+ if (a->hangover > 1) {
+ --(a->hangover);
+ } else if (1 == a->hangover) {
+ --(a->hangover);
+ // My Leaky NLMS is to erase vector w when hangover expires
+ memset(a->w, 0, sizeof(a->w));
+ }
+ }
+}
+
+
+#if 0
+void AEC::openwdisplay() {
+ // open TCP connection to program wdisplay.tcl
+ fdwdisplay = socket_async("127.0.0.1", 50999);
+};
+#endif
+
+
+static REAL AEC_nlms_pw(AEC *a, REAL d, REAL x_, float stepsize)
+{
+ REAL e;
+ REAL ef;
+ a->x[a->j] = x_;
+ a->xf[a->j] = IIR1_highpass(a->Fx, x_); // pre-whitening of x
+
+ // calculate error value
+ // (mic signal - estimated mic signal from spk signal)
+ e = d;
+ if (a->hangover > 0) {
+ e -= a->dotp(a->w, a->x + a->j);
+ }
+ ef = IIR1_highpass(a->Fe, e); // pre-whitening of e
+
+ // optimize: iterative dotp(xf, xf)
+ a->dotp_xf_xf += (a->xf[a->j] * a->xf[a->j] - a->xf[a->j + NLMS_LEN - 1] * a->xf[a->j + NLMS_LEN - 1]);
+
+ if (stepsize > 0.0) {
+ // calculate variable step size
+ REAL mikro_ef = stepsize * ef / a->dotp_xf_xf;
+
+#ifdef DISABLE_ORC
+ // update tap weights (filter learning)
+ int i;
+ for (i = 0; i < NLMS_LEN; i += 2) {
+ // optimize: partial loop unrolling
+ a->w[i] += mikro_ef * a->xf[i + a->j];
+ a->w[i + 1] += mikro_ef * a->xf[i + a->j + 1];
+ }
+#else
+ update_tap_weights(a->w, &a->xf[a->j], mikro_ef, NLMS_LEN);
+#endif
+ }
+
+ if (--(a->j) < 0) {
+ // optimize: decrease number of memory copies
+ a->j = NLMS_EXT;
+ memmove(a->x + a->j + 1, a->x, (NLMS_LEN - 1) * sizeof(REAL));
+ memmove(a->xf + a->j + 1, a->xf, (NLMS_LEN - 1) * sizeof(REAL));
+ }
+
+ // Saturation
+ if (e > MAXPCM) {
+ return MAXPCM;
+ } else if (e < -MAXPCM) {
+ return -MAXPCM;
+ } else {
+ return e;
+ }
+}
+
+
+int AEC_doAEC(AEC *a, int d_, int x_)
+{
+ REAL d = (REAL) d_;
+ REAL x = (REAL) x_;
+
+ // Mic Highpass Filter - to remove DC
+ d = IIR_HP_highpass(a->acMic, d);
+
+ // Mic Highpass Filter - cut-off below 300Hz
+ d = FIR_HP_300Hz_highpass(a->cutoff, d);
+
+ // Amplify, for e.g. Soundcards with -6dB max. volume
+ d *= a->gain;
+
+ // Spk Highpass Filter - to remove DC
+ x = IIR_HP_highpass(a->acSpk, x);
+
+ // Double Talk Detector
+ a->stepsize = AEC_dtd(a, d, x);
+
+ // Leaky (ageing of vector w)
+ AEC_leaky(a);
+
+ // Acoustic Echo Cancellation
+ d = AEC_nlms_pw(a, d, x, a->stepsize);
+
+#if 0
+ if (fdwdisplay >= 0) {
+ if (++dumpcnt >= (WIDEB*RATE/10)) {
+ // wdisplay creates 10 dumps per seconds = large CPU load!
+ dumpcnt = 0;
+ write(fdwdisplay, ws, DUMP_LEN*sizeof(float));
+ // we don't check return value. This is not production quality!!!
+ memset(ws, 0, sizeof(ws));
+ } else {
+ int i;
+ for (i = 0; i < DUMP_LEN; i += 2) {
+ // optimize: partial loop unrolling
+ ws[i] += w[i];
+ ws[i + 1] += w[i + 1];
+ }
+ }
+ }
+#endif
+
+ return (int) d;
+}
diff --git a/src/modules/echo-cancel/adrian-aec.h b/src/modules/echo-cancel/adrian-aec.h
new file mode 100644
index 00000000..d024b3c5
--- /dev/null
+++ b/src/modules/echo-cancel/adrian-aec.h
@@ -0,0 +1,382 @@
+/* aec.h
+ *
+ * Copyright (C) DFS Deutsche Flugsicherung (2004, 2005).
+ * All Rights Reserved.
+ * Author: Andre Adrian
+ *
+ * Acoustic Echo Cancellation Leaky NLMS-pw algorithm
+ *
+ * Version 0.3 filter created with www.dsptutor.freeuk.com
+ * Version 0.3.1 Allow change of stability parameter delta
+ * Version 0.4 Leaky Normalized LMS - pre whitening algorithm
+ */
+
+#ifndef _AEC_H /* include only once */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/gccmacro.h>
+#include <pulse/xmalloc.h>
+
+#include <pulsecore/macro.h>
+
+#define WIDEB 2
+
+// use double if your CPU does software-emulation of float
+#define REAL float
+
+/* dB Values */
+#define M0dB 1.0f
+#define M3dB 0.71f
+#define M6dB 0.50f
+#define M9dB 0.35f
+#define M12dB 0.25f
+#define M18dB 0.125f
+#define M24dB 0.063f
+
+/* dB values for 16bit PCM */
+/* MxdB_PCM = 32767 * 10 ^(x / 20) */
+#define M10dB_PCM 10362.0f
+#define M20dB_PCM 3277.0f
+#define M25dB_PCM 1843.0f
+#define M30dB_PCM 1026.0f
+#define M35dB_PCM 583.0f
+#define M40dB_PCM 328.0f
+#define M45dB_PCM 184.0f
+#define M50dB_PCM 104.0f
+#define M55dB_PCM 58.0f
+#define M60dB_PCM 33.0f
+#define M65dB_PCM 18.0f
+#define M70dB_PCM 10.0f
+#define M75dB_PCM 6.0f
+#define M80dB_PCM 3.0f
+#define M85dB_PCM 2.0f
+#define M90dB_PCM 1.0f
+
+#define MAXPCM 32767.0f
+
+/* Design constants (Change to fine tune the algorithms */
+
+/* The following values are for hardware AEC and studio quality
+ * microphone */
+
+/* NLMS filter length in taps (samples). A longer filter length gives
+ * better Echo Cancellation, but maybe slower convergence speed and
+ * needs more CPU power (Order of NLMS is linear) */
+#define NLMS_LEN (100*WIDEB*8)
+
+/* Vector w visualization length in taps (samples).
+ * Must match argv value for wdisplay.tcl */
+#define DUMP_LEN (40*WIDEB*8)
+
+/* minimum energy in xf. Range: M70dB_PCM to M50dB_PCM. Should be equal
+ * to microphone ambient Noise level */
+#define NoiseFloor M55dB_PCM
+
+/* Leaky hangover in taps.
+ */
+#define Thold (60 * WIDEB * 8)
+
+// Adrian soft decision DTD
+// left point. X is ratio, Y is stepsize
+#define STEPX1 1.0
+#define STEPY1 1.0
+// right point. STEPX2=2.0 is good double talk, 3.0 is good single talk.
+#define STEPX2 2.5
+#define STEPY2 0
+#define ALPHAFAST (1.0f / 100.0f)
+#define ALPHASLOW (1.0f / 20000.0f)
+
+
+
+/* Ageing multiplier for LMS memory vector w */
+#define Leaky 0.9999f
+
+/* Double Talk Detector Speaker/Microphone Threshold. Range <=1
+ * Large value (M0dB) is good for Single-Talk Echo cancellation,
+ * small value (M12dB) is good for Doulbe-Talk AEC */
+#define GeigelThreshold M6dB
+
+/* for Non Linear Processor. Range >0 to 1. Large value (M0dB) is good
+ * for Double-Talk, small value (M12dB) is good for Single-Talk */
+#define NLPAttenuation M12dB
+
+/* Below this line there are no more design constants */
+
+typedef struct IIR_HP IIR_HP;
+
+/* Exponential Smoothing or IIR Infinite Impulse Response Filter */
+struct IIR_HP {
+ REAL x;
+};
+
+static IIR_HP* IIR_HP_init(void) {
+ IIR_HP *i = pa_xnew(IIR_HP, 1);
+ i->x = 0.0f;
+ return i;
+ }
+
+static REAL IIR_HP_highpass(IIR_HP *i, REAL in) {
+ const REAL a0 = 0.01f; /* controls Transfer Frequency */
+ /* Highpass = Signal - Lowpass. Lowpass = Exponential Smoothing */
+ i->x += a0 * (in - i->x);
+ return in - i->x;
+ };
+
+typedef struct FIR_HP_300Hz FIR_HP_300Hz;
+
+#if WIDEB==1
+/* 17 taps FIR Finite Impulse Response filter
+ * Coefficients calculated with
+ * www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
+ */
+class FIR_HP_300Hz {
+ REAL z[18];
+
+public:
+ FIR_HP_300Hz() {
+ memset(this, 0, sizeof(FIR_HP_300Hz));
+ }
+
+ REAL highpass(REAL in) {
+ const REAL a[18] = {
+ // Kaiser Window FIR Filter, Filter type: High pass
+ // Passband: 300.0 - 4000.0 Hz, Order: 16
+ // Transition band: 75.0 Hz, Stopband attenuation: 10.0 dB
+ -0.034870606, -0.039650206, -0.044063766, -0.04800318,
+ -0.051370874, -0.054082647, -0.056070227, -0.057283327,
+ 0.8214126, -0.057283327, -0.056070227, -0.054082647,
+ -0.051370874, -0.04800318, -0.044063766, -0.039650206,
+ -0.034870606, 0.0
+ };
+ memmove(z + 1, z, 17 * sizeof(REAL));
+ z[0] = in;
+ REAL sum0 = 0.0, sum1 = 0.0;
+ int j;
+
+ for (j = 0; j < 18; j += 2) {
+ // optimize: partial loop unrolling
+ sum0 += a[j] * z[j];
+ sum1 += a[j + 1] * z[j + 1];
+ }
+ return sum0 + sum1;
+ }
+};
+
+#else
+
+/* 35 taps FIR Finite Impulse Response filter
+ * Passband 150Hz to 4kHz for 8kHz sample rate, 300Hz to 8kHz for 16kHz
+ * sample rate.
+ * Coefficients calculated with
+ * www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
+ */
+struct FIR_HP_300Hz {
+ REAL z[36];
+};
+
+static FIR_HP_300Hz* FIR_HP_300Hz_init(void) {
+ FIR_HP_300Hz *ret = pa_xnew(FIR_HP_300Hz, 1);
+ memset(ret, 0, sizeof(FIR_HP_300Hz));
+ return ret;
+ }
+
+static REAL FIR_HP_300Hz_highpass(FIR_HP_300Hz *f, REAL in) {
+ REAL sum0 = 0.0, sum1 = 0.0;
+ int j;
+ const REAL a[36] = {
+ // Kaiser Window FIR Filter, Filter type: High pass
+ // Passband: 150.0 - 4000.0 Hz, Order: 34
+ // Transition band: 34.0 Hz, Stopband attenuation: 10.0 dB
+ -0.016165324, -0.017454365, -0.01871232, -0.019931411,
+ -0.021104068, -0.022222936, -0.02328091, -0.024271343,
+ -0.025187887, -0.02602462, -0.026776174, -0.027437767,
+ -0.028004972, -0.028474221, -0.028842418, -0.029107114,
+ -0.02926664, 0.8524841, -0.02926664, -0.029107114,
+ -0.028842418, -0.028474221, -0.028004972, -0.027437767,
+ -0.026776174, -0.02602462, -0.025187887, -0.024271343,
+ -0.02328091, -0.022222936, -0.021104068, -0.019931411,
+ -0.01871232, -0.017454365, -0.016165324, 0.0
+ };
+ memmove(f->z + 1, f->z, 35 * sizeof(REAL));
+ f->z[0] = in;
+
+ for (j = 0; j < 36; j += 2) {
+ // optimize: partial loop unrolling
+ sum0 += a[j] * f->z[j];
+ sum1 += a[j + 1] * f->z[j + 1];
+ }
+ return sum0 + sum1;
+ }
+#endif
+
+typedef struct IIR1 IIR1;
+
+/* Recursive single pole IIR Infinite Impulse response High-pass filter
+ *
+ * Reference: The Scientist and Engineer's Guide to Digital Processing
+ *
+ * output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
+ *
+ * X = exp(-2.0 * pi * Fc)
+ * A0 = (1 + X) / 2
+ * A1 = -(1 + X) / 2
+ * B1 = X
+ * Fc = cutoff freq / sample rate
+ */
+struct IIR1 {
+ REAL in0, out0;
+ REAL a0, a1, b1;
+};
+
+#if 0
+ IIR1() {
+ memset(this, 0, sizeof(IIR1));
+ }
+#endif
+
+static IIR1* IIR1_init(REAL Fc) {
+ IIR1 *i = pa_xnew(IIR1, 1);
+ i->b1 = expf(-2.0f * M_PI * Fc);
+ i->a0 = (1.0f + i->b1) / 2.0f;
+ i->a1 = -(i->a0);
+ i->in0 = 0.0f;
+ i->out0 = 0.0f;
+ return i;
+ }
+
+static REAL IIR1_highpass(IIR1 *i, REAL in) {
+ REAL out = i->a0 * in + i->a1 * i->in0 + i->b1 * i->out0;
+ i->in0 = in;
+ i->out0 = out;
+ return out;
+ }
+
+
+#if 0
+/* Recursive two pole IIR Infinite Impulse Response filter
+ * Coefficients calculated with
+ * http://www.dsptutor.freeuk.com/IIRFilterDesign/IIRFiltDes102.html
+ */
+class IIR2 {
+ REAL x[2], y[2];
+
+public:
+ IIR2() {
+ memset(this, 0, sizeof(IIR2));
+ }
+
+ REAL highpass(REAL in) {
+ // Butterworth IIR filter, Filter type: HP
+ // Passband: 2000 - 4000.0 Hz, Order: 2
+ const REAL a[] = { 0.29289323f, -0.58578646f, 0.29289323f };
+ const REAL b[] = { 1.3007072E-16f, 0.17157288f };
+ REAL out =
+ a[0] * in + a[1] * x[0] + a[2] * x[1] - b[0] * y[0] - b[1] * y[1];
+
+ x[1] = x[0];
+ x[0] = in;
+ y[1] = y[0];
+ y[0] = out;
+ return out;
+ }
+};
+#endif
+
+
+// Extention in taps to reduce mem copies
+#define NLMS_EXT (10*8)
+
+// block size in taps to optimize DTD calculation
+#define DTD_LEN 16
+
+typedef struct AEC AEC;
+
+struct AEC {
+ // Time domain Filters
+ IIR_HP *acMic, *acSpk; // DC-level remove Highpass)
+ FIR_HP_300Hz *cutoff; // 150Hz cut-off Highpass
+ REAL gain; // Mic signal amplify
+ IIR1 *Fx, *Fe; // pre-whitening Highpass for x, e
+
+ // Adrian soft decision DTD (Double Talk Detector)
+ REAL dfast, xfast;
+ REAL dslow, xslow;
+
+ // NLMS-pw
+ REAL x[NLMS_LEN + NLMS_EXT]; // tap delayed loudspeaker signal
+ REAL xf[NLMS_LEN + NLMS_EXT]; // pre-whitening tap delayed signal
+ REAL w_arr[NLMS_LEN + (16 / sizeof(REAL))]; // tap weights
+ REAL *w; // this will be a 16-byte aligned pointer into w_arr
+ int j; // optimize: less memory copies
+ double dotp_xf_xf; // double to avoid loss of precision
+ float delta; // noise floor to stabilize NLMS
+
+ // AES
+ float aes_y2; // not in use!
+
+ // w vector visualization
+ REAL ws[DUMP_LEN]; // tap weights sums
+ int fdwdisplay; // TCP file descriptor
+ int dumpcnt; // wdisplay output counter
+
+ // variables are public for visualization
+ int hangover;
+ float stepsize;
+
+ // vfuncs that are picked based on processor features available
+ REAL (*dotp) (REAL[], REAL[]);
+};
+
+/* Double-Talk Detector
+ *
+ * in d: microphone sample (PCM as REALing point value)
+ * in x: loudspeaker sample (PCM as REALing point value)
+ * return: from 0 for doubletalk to 1.0 for single talk
+ */
+static float AEC_dtd(AEC *a, REAL d, REAL x);
+
+static void AEC_leaky(AEC *a);
+
+/* Normalized Least Mean Square Algorithm pre-whitening (NLMS-pw)
+ * The LMS algorithm was developed by Bernard Widrow
+ * book: Haykin, Adaptive Filter Theory, 4. edition, Prentice Hall, 2002
+ *
+ * in d: microphone sample (16bit PCM value)
+ * in x_: loudspeaker sample (16bit PCM value)
+ * in stepsize: NLMS adaptation variable
+ * return: echo cancelled microphone sample
+ */
+static REAL AEC_nlms_pw(AEC *a, REAL d, REAL x_, float stepsize);
+
+ AEC* AEC_init(int RATE, int have_vector);
+
+/* Acoustic Echo Cancellation and Suppression of one sample
+ * in d: microphone signal with echo
+ * in x: loudspeaker signal
+ * return: echo cancelled microphone signal
+ */
+ int AEC_doAEC(AEC *a, int d_, int x_);
+
+PA_GCC_UNUSED static float AEC_getambient(AEC *a) {
+ return a->dfast;
+ };
+static void AEC_setambient(AEC *a, float Min_xf) {
+ a->dotp_xf_xf -= a->delta; // subtract old delta
+ a->delta = (NLMS_LEN-1) * Min_xf * Min_xf;
+ a->dotp_xf_xf += a->delta; // add new delta
+ };
+PA_GCC_UNUSED static void AEC_setgain(AEC *a, float gain_) {
+ a->gain = gain_;
+ };
+#if 0
+ void AEC_openwdisplay(AEC *a);
+#endif
+PA_GCC_UNUSED static void AEC_setaes(AEC *a, float aes_y2_) {
+ a->aes_y2 = aes_y2_;
+ };
+
+#define _AEC_H
+#endif
diff --git a/src/modules/echo-cancel/adrian-aec.orc b/src/modules/echo-cancel/adrian-aec.orc
new file mode 100644
index 00000000..80547723
--- /dev/null
+++ b/src/modules/echo-cancel/adrian-aec.orc
@@ -0,0 +1,8 @@
+.function update_tap_weights
+.dest 4 w float
+.source 4 xf float
+.floatparam 4 mikro_ef
+.temp 4 tmp float
+
+mulf tmp, mikro_ef, xf
+addf w, w, tmp
diff --git a/src/modules/echo-cancel/adrian-license.txt b/src/modules/echo-cancel/adrian-license.txt
new file mode 100644
index 00000000..7c06efd0
--- /dev/null
+++ b/src/modules/echo-cancel/adrian-license.txt
@@ -0,0 +1,17 @@
+ Copyright (C) DFS Deutsche Flugsicherung (2004). All Rights Reserved.
+
+ You are allowed to use this source code in any open source or closed
+ source software you want. You are allowed to use the algorithms for a
+ hardware solution. You are allowed to modify the source code.
+ You are not allowed to remove the name of the author from this memo or
+ from the source code files. You are not allowed to monopolize the
+ source code or the algorithms behind the source code as your
+ intellectual property. This source code is free of royalty and comes
+ with no warranty.
+
+--- The following does not apply to the PulseAudio module ---
+
+ Please see g711/gen-lic.txt for the ITU-T G.711 codec copyright.
+ Please see gsm/gen-lic.txt for the ITU-T GSM codec copyright.
+ Please see ilbc/COPYRIGHT and ilbc/NOTICE for the IETF iLBC codec
+ copyright.
diff --git a/src/modules/echo-cancel/adrian.c b/src/modules/echo-cancel/adrian.c
new file mode 100644
index 00000000..ab3858a4
--- /dev/null
+++ b/src/modules/echo-cancel/adrian.c
@@ -0,0 +1,117 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2010 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ Contributor: Wim Taymans <wim.taymans@gmail.com>
+
+ The actual implementation is taken from the sources at
+ http://andreadrian.de/intercom/ - for the license, look for
+ adrian-license.txt in the same directory as this file.
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/xmalloc.h>
+
+#include <pulsecore/modargs.h>
+
+#include "echo-cancel.h"
+
+/* should be between 10-20 ms */
+#define DEFAULT_FRAME_SIZE_MS 20
+
+static const char* const valid_modargs[] = {
+ "frame_size_ms",
+ NULL
+};
+
+static void pa_adrian_ec_fixate_spec(pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map)
+{
+ source_ss->format = PA_SAMPLE_S16NE;
+ source_ss->channels = 1;
+ pa_channel_map_init_mono(source_map);
+
+ *sink_ss = *source_ss;
+ *sink_map = *source_map;
+}
+
+pa_bool_t pa_adrian_ec_init(pa_core *c, pa_echo_canceller *ec,
+ pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map,
+ uint32_t *blocksize, const char *args)
+{
+ int framelen, rate, have_vector = 0;
+ uint32_t frame_size_ms;
+ pa_modargs *ma;
+
+ if (!(ma = pa_modargs_new(args, valid_modargs))) {
+ pa_log("Failed to parse submodule arguments.");
+ goto fail;
+ }
+
+ frame_size_ms = DEFAULT_FRAME_SIZE_MS;
+ if (pa_modargs_get_value_u32(ma, "frame_size_ms", &frame_size_ms) < 0 || frame_size_ms < 1 || frame_size_ms > 200) {
+ pa_log("Invalid frame_size_ms specification");
+ goto fail;
+ }
+
+ pa_adrian_ec_fixate_spec(source_ss, source_map, sink_ss, sink_map);
+
+ rate = source_ss->rate;
+ framelen = (rate * frame_size_ms) / 1000;
+
+ *blocksize = ec->params.priv.adrian.blocksize = framelen * pa_frame_size (source_ss);
+
+ pa_log_debug ("Using framelen %d, blocksize %u, channels %d, rate %d", framelen, ec->params.priv.adrian.blocksize, source_ss->channels, source_ss->rate);
+
+ /* For now we only support SSE */
+ if (c->cpu_info.cpu_type == PA_CPU_X86 && (c->cpu_info.flags.x86 & PA_CPU_X86_SSE))
+ have_vector = 1;
+
+ ec->params.priv.adrian.aec = AEC_init(rate, have_vector);
+ if (!ec->params.priv.adrian.aec)
+ goto fail;
+
+ pa_modargs_free(ma);
+ return TRUE;
+
+fail:
+ if (ma)
+ pa_modargs_free(ma);
+ return FALSE;
+}
+
+void pa_adrian_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
+ unsigned int i;
+
+ for (i = 0; i < ec->params.priv.adrian.blocksize; i += 2) {
+ /* We know it's S16NE mono data */
+ int r = *(int16_t *)(rec + i);
+ int p = *(int16_t *)(play + i);
+ *(int16_t *)(out + i) = (int16_t) AEC_doAEC(ec->params.priv.adrian.aec, r, p);
+ }
+}
+
+void pa_adrian_ec_done(pa_echo_canceller *ec) {
+ pa_xfree(ec->params.priv.adrian.aec);
+ ec->params.priv.adrian.aec = NULL;
+}
diff --git a/src/modules/echo-cancel/adrian.h b/src/modules/echo-cancel/adrian.h
new file mode 100644
index 00000000..639fa9ec
--- /dev/null
+++ b/src/modules/echo-cancel/adrian.h
@@ -0,0 +1,31 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2010 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ The actual implementation is taken from the sources at
+ http://andreadrian.de/intercom/ - for the license, look for
+ adrian-license.txt in the same directory as this file.
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+/* Forward declarations */
+
+typedef struct AEC AEC;
+
+AEC* AEC_init(int RATE, int have_vector);
+int AEC_doAEC(AEC *a, int d_, int x_);
diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h
new file mode 100644
index 00000000..aa40adce
--- /dev/null
+++ b/src/modules/echo-cancel/echo-cancel.h
@@ -0,0 +1,90 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2010 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/sample.h>
+#include <pulse/channelmap.h>
+#include <pulsecore/core.h>
+#include <pulsecore/macro.h>
+
+#include <speex/speex_echo.h>
+#include <speex/speex_preprocess.h>
+#include "adrian.h"
+
+/* Common data structures */
+
+typedef struct pa_echo_canceller_params pa_echo_canceller_params;
+
+struct pa_echo_canceller_params {
+ union {
+ struct {
+ SpeexEchoState *state;
+ } speex;
+ struct {
+ uint32_t blocksize;
+ AEC *aec;
+ } adrian;
+ /* each canceller-specific structure goes here */
+ } priv;
+};
+
+typedef struct pa_echo_canceller pa_echo_canceller;
+
+struct pa_echo_canceller {
+ pa_bool_t (*init) (pa_core *c,
+ pa_echo_canceller *ec,
+ pa_sample_spec *source_ss,
+ pa_channel_map *source_map,
+ pa_sample_spec *sink_ss,
+ pa_channel_map *sink_map,
+ uint32_t *blocksize,
+ const char *args);
+ void (*run) (pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out);
+ void (*done) (pa_echo_canceller *ec);
+
+ pa_echo_canceller_params params;
+
+ pa_bool_t agc;
+ pa_bool_t denoise;
+ pa_bool_t echo_suppress;
+ int32_t echo_suppress_attenuation;
+ int32_t echo_suppress_attenuation_active;
+ SpeexPreprocessState *pp_state;
+};
+
+/* Speex canceller functions */
+pa_bool_t pa_speex_ec_init(pa_core *c, pa_echo_canceller *ec,
+ pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map,
+ uint32_t *blocksize, const char *args);
+void pa_speex_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out);
+void pa_speex_ec_done(pa_echo_canceller *ec);
+
+/* Adrian Andre's echo canceller */
+pa_bool_t pa_adrian_ec_init(pa_core *c, pa_echo_canceller *ec,
+ pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map,
+ uint32_t *blocksize, const char *args);
+void pa_adrian_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out);
+void pa_adrian_ec_done(pa_echo_canceller *ec);
diff --git a/src/modules/echo-cancel/module-echo-cancel.c b/src/modules/echo-cancel/module-echo-cancel.c
new file mode 100644
index 00000000..b84bf1db
--- /dev/null
+++ b/src/modules/echo-cancel/module-echo-cancel.c
@@ -0,0 +1,1778 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2010 Wim Taymans <wim.taymans@gmail.com>
+
+ Based on module-virtual-sink.c
+ module-virtual-source.c
+ module-loopback.c
+
+ Copyright 2010 Intel Corporation
+ Contributor: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdio.h>
+
+#include "echo-cancel.h"
+
+#include <pulse/xmalloc.h>
+#include <pulse/i18n.h>
+#include <pulse/timeval.h>
+#include <pulse/rtclock.h>
+
+#include <pulsecore/atomic.h>
+#include <pulsecore/macro.h>
+#include <pulsecore/namereg.h>
+#include <pulsecore/sink.h>
+#include <pulsecore/module.h>
+#include <pulsecore/core-rtclock.h>
+#include <pulsecore/core-util.h>
+#include <pulsecore/modargs.h>
+#include <pulsecore/log.h>
+#include <pulsecore/rtpoll.h>
+#include <pulsecore/sample-util.h>
+#include <pulsecore/ltdl-helper.h>
+
+#include "module-echo-cancel-symdef.h"
+
+PA_MODULE_AUTHOR("Wim Taymans");
+PA_MODULE_DESCRIPTION("Echo Cancelation");
+PA_MODULE_VERSION(PACKAGE_VERSION);
+PA_MODULE_LOAD_ONCE(FALSE);
+PA_MODULE_USAGE(
+ _("source_name=<name for the source> "
+ "source_properties=<properties for the source> "
+ "source_master=<name of source to filter> "
+ "sink_name=<name for the sink> "
+ "sink_properties=<properties for the sink> "
+ "sink_master=<name of sink to filter> "
+ "adjust_time=<how often to readjust rates in s> "
+ "format=<sample format> "
+ "rate=<sample rate> "
+ "channels=<number of channels> "
+ "channel_map=<channel map> "
+ "aec_method=<implementation to use> "
+ "aec_args=<parameters for the AEC engine> "
+ "agc=<perform automagic gain control?> "
+ "denoise=<apply denoising?> "
+ "echo_suppress=<perform residual echo suppression? (only with the speex canceller)> "
+ "echo_suppress_attenuation=<dB value of residual echo attenuation> "
+ "echo_suppress_attenuation_active=<dB value of residual echo attenuation when near end is active> "
+ "save_aec=<save AEC data in /tmp> "
+ "autoloaded=<set if this module is being loaded automatically> "
+ ));
+
+/* NOTE: Make sure the enum and ec_table are maintained in the correct order */
+typedef enum {
+ PA_ECHO_CANCELLER_INVALID = -1,
+ PA_ECHO_CANCELLER_SPEEX = 0,
+ PA_ECHO_CANCELLER_ADRIAN,
+} pa_echo_canceller_method_t;
+
+#define DEFAULT_ECHO_CANCELLER "speex"
+
+static const pa_echo_canceller ec_table[] = {
+ {
+ /* Speex */
+ .init = pa_speex_ec_init,
+ .run = pa_speex_ec_run,
+ .done = pa_speex_ec_done,
+ },
+ {
+ /* Adrian Andre's NLMS implementation */
+ .init = pa_adrian_ec_init,
+ .run = pa_adrian_ec_run,
+ .done = pa_adrian_ec_done,
+ },
+};
+
+#define DEFAULT_ADJUST_TIME_USEC (1*PA_USEC_PER_SEC)
+#define DEFAULT_AGC_ENABLED FALSE
+#define DEFAULT_DENOISE_ENABLED FALSE
+#define DEFAULT_ECHO_SUPPRESS_ENABLED FALSE
+#define DEFAULT_ECHO_SUPPRESS_ATTENUATION 0
+#define DEFAULT_SAVE_AEC 0
+#define DEFAULT_AUTOLOADED FALSE
+
+#define MEMBLOCKQ_MAXLENGTH (16*1024*1024)
+
+/* This module creates a new (virtual) source and sink.
+ *
+ * The data sent to the new sink is kept in a memblockq before being
+ * forwarded to the real sink_master.
+ *
+ * Data read from source_master is matched against the saved sink data and
+ * echo canceled data is then pushed onto the new source.
+ *
+ * Both source and sink masters have their own threads to push/pull data
+ * respectively. We however perform all our actions in the source IO thread.
+ * To do this we send all played samples to the source IO thread where they
+ * are then pushed into the memblockq.
+ *
+ * Alignment is performed in two steps:
+ *
+ * 1) when something happens that requires quick adjustement of the alignment of
+ * capture and playback samples, we perform a resync. This adjusts the
+ * position in the playback memblock to the requested sample. Quick
+ * adjustements include moving the playback samples before the capture
+ * samples (because else the echo canceler does not work) or when the
+ * playback pointer drifts too far away.
+ *
+ * 2) periodically check the difference between capture and playback. we use a
+ * low and high watermark for adjusting the alignment. playback should always
+ * be before capture and the difference should not be bigger than one frame
+ * size. We would ideally like to resample the sink_input but most driver
+ * don't give enough accuracy to be able to do that right now.
+ */
+
+struct snapshot {
+ pa_usec_t sink_now;
+ pa_usec_t sink_latency;
+ size_t sink_delay;
+ int64_t send_counter;
+
+ pa_usec_t source_now;
+ pa_usec_t source_latency;
+ size_t source_delay;
+ int64_t recv_counter;
+ size_t rlen;
+ size_t plen;
+};
+
+struct userdata {
+ pa_core *core;
+ pa_module *module;
+
+ pa_bool_t autoloaded;
+ uint32_t save_aec;
+
+ pa_echo_canceller *ec;
+ uint32_t blocksize;
+
+ pa_bool_t need_realign;
+
+ /* to wakeup the source I/O thread */
+ pa_bool_t in_push;
+ pa_asyncmsgq *asyncmsgq;
+ pa_rtpoll_item *rtpoll_item_read, *rtpoll_item_write;
+
+ pa_source *source;
+ pa_bool_t source_auto_desc;
+ pa_source_output *source_output;
+ pa_memblockq *source_memblockq; /* echo canceler needs fixed sized chunks */
+ size_t source_skip;
+
+ pa_sink *sink;
+ pa_bool_t sink_auto_desc;
+ pa_sink_input *sink_input;
+ pa_memblockq *sink_memblockq;
+ int64_t send_counter; /* updated in sink IO thread */
+ int64_t recv_counter;
+ size_t sink_skip;
+
+ pa_atomic_t request_resync;
+
+ int active_mask;
+ pa_time_event *time_event;
+ pa_usec_t adjust_time;
+
+ FILE *captured_file;
+ FILE *played_file;
+ FILE *canceled_file;
+};
+
+static void source_output_snapshot_within_thread(struct userdata *u, struct snapshot *snapshot);
+
+static const char* const valid_modargs[] = {
+ "source_name",
+ "source_properties",
+ "source_master",
+ "sink_name",
+ "sink_properties",
+ "sink_master",
+ "adjust_time",
+ "format",
+ "rate",
+ "channels",
+ "channel_map",
+ "aec_method",
+ "aec_args",
+ "agc",
+ "denoise",
+ "echo_suppress",
+ "echo_suppress_attenuation",
+ "echo_suppress_attenuation_active",
+ "save_aec",
+ "autoloaded",
+ NULL
+};
+
+enum {
+ SOURCE_OUTPUT_MESSAGE_POST = PA_SOURCE_OUTPUT_MESSAGE_MAX,
+ SOURCE_OUTPUT_MESSAGE_REWIND,
+ SOURCE_OUTPUT_MESSAGE_LATENCY_SNAPSHOT,
+ SOURCE_OUTPUT_MESSAGE_APPLY_DIFF_TIME
+};
+
+enum {
+ SINK_INPUT_MESSAGE_LATENCY_SNAPSHOT
+};
+
+static int64_t calc_diff(struct userdata *u, struct snapshot *snapshot) {
+ int64_t buffer, diff_time, buffer_latency;
+
+ /* get the number of samples between capture and playback */
+ if (snapshot->plen > snapshot->rlen)
+ buffer = snapshot->plen - snapshot->rlen;
+ else
+ buffer = 0;
+
+ buffer += snapshot->source_delay + snapshot->sink_delay;
+
+ /* add the amount of samples not yet transfered to the source context */
+ if (snapshot->recv_counter <= snapshot->send_counter)
+ buffer += (int64_t) (snapshot->send_counter - snapshot->recv_counter);
+ else
+ buffer += PA_CLIP_SUB(buffer, (int64_t) (snapshot->recv_counter - snapshot->send_counter));
+
+ /* convert to time */
+ buffer_latency = pa_bytes_to_usec(buffer, &u->source_output->sample_spec);
+
+ /* capture and playback samples are perfectly aligned when diff_time is 0 */
+ diff_time = (snapshot->sink_now + snapshot->sink_latency - buffer_latency) -
+ (snapshot->source_now - snapshot->source_latency);
+
+ pa_log_debug("diff %lld (%lld - %lld + %lld) %lld %lld %lld %lld", (long long) diff_time,
+ (long long) snapshot->sink_latency,
+ (long long) buffer_latency, (long long) snapshot->source_latency,
+ (long long) snapshot->source_delay, (long long) snapshot->sink_delay,
+ (long long) (snapshot->send_counter - snapshot->recv_counter),
+ (long long) (snapshot->sink_now - snapshot->source_now));
+
+ return diff_time;
+}
+
+/* Called from main context */
+static void time_callback(pa_mainloop_api *a, pa_time_event *e, const struct timeval *t, void *userdata) {
+ struct userdata *u = userdata;
+ uint32_t old_rate, base_rate, new_rate;
+ int64_t diff_time;
+ /*size_t fs*/
+ struct snapshot latency_snapshot;
+
+ pa_assert(u);
+ pa_assert(a);
+ pa_assert(u->time_event == e);
+ pa_assert_ctl_context();
+
+ if (u->active_mask != 3)
+ return;
+
+ /* update our snapshots */
+ pa_asyncmsgq_send(u->source_output->source->asyncmsgq, PA_MSGOBJECT(u->source_output), SOURCE_OUTPUT_MESSAGE_LATENCY_SNAPSHOT, &latency_snapshot, 0, NULL);
+ pa_asyncmsgq_send(u->sink_input->sink->asyncmsgq, PA_MSGOBJECT(u->sink_input), SINK_INPUT_MESSAGE_LATENCY_SNAPSHOT, &latency_snapshot, 0, NULL);
+
+ /* calculate drift between capture and playback */
+ diff_time = calc_diff(u, &latency_snapshot);
+
+ /*fs = pa_frame_size(&u->source_output->sample_spec);*/
+ old_rate = u->sink_input->sample_spec.rate;
+ base_rate = u->source_output->sample_spec.rate;
+
+ if (diff_time < 0) {
+ /* recording before playback, we need to adjust quickly. The echo
+ * canceler does not work in this case. */
+ pa_asyncmsgq_post(u->asyncmsgq, PA_MSGOBJECT(u->source_output), SOURCE_OUTPUT_MESSAGE_APPLY_DIFF_TIME,
+ NULL, diff_time, NULL, NULL);
+ /*new_rate = base_rate - ((pa_usec_to_bytes(-diff_time, &u->source_output->sample_spec) / fs) * PA_USEC_PER_SEC) / u->adjust_time;*/
+ new_rate = base_rate;
+ }
+ else {
+ if (diff_time > 1000) {
+ /* diff too big, quickly adjust */
+ pa_asyncmsgq_post(u->asyncmsgq, PA_MSGOBJECT(u->source_output), SOURCE_OUTPUT_MESSAGE_APPLY_DIFF_TIME,
+ NULL, diff_time, NULL, NULL);
+ }
+
+ /* recording behind playback, we need to slowly adjust the rate to match */
+ /*new_rate = base_rate + ((pa_usec_to_bytes(diff_time, &u->source_output->sample_spec) / fs) * PA_USEC_PER_SEC) / u->adjust_time;*/
+
+ /* assume equal samplerates for now */
+ new_rate = base_rate;
+ }
+
+ /* make sure we don't make too big adjustements because that sounds horrible */
+ if (new_rate > base_rate * 1.1 || new_rate < base_rate * 0.9)
+ new_rate = base_rate;
+
+ if (new_rate != old_rate) {
+ pa_log_info("Old rate %lu Hz, new rate %lu Hz", (unsigned long) old_rate, (unsigned long) new_rate);
+
+ pa_sink_input_set_rate(u->sink_input, new_rate);
+ }
+
+ pa_core_rttime_restart(u->core, u->time_event, pa_rtclock_now() + u->adjust_time);
+}
+
+/* Called from source I/O thread context */
+static int source_process_msg_cb(pa_msgobject *o, int code, void *data, int64_t offset, pa_memchunk *chunk) {
+ struct userdata *u = PA_SOURCE(o)->userdata;
+
+ switch (code) {
+
+ case PA_SOURCE_MESSAGE_GET_LATENCY:
+
+ /* The source is _put() before the source output is, so let's
+ * make sure we don't access it in that time. Also, the
+ * source output is first shut down, the source second. */
+ if (!PA_SOURCE_IS_LINKED(u->source->thread_info.state) ||
+ !PA_SOURCE_OUTPUT_IS_LINKED(u->source_output->thread_info.state)) {
+ *((pa_usec_t*) data) = 0;
+ return 0;
+ }
+
+ *((pa_usec_t*) data) =
+
+ /* Get the latency of the master source */
+ pa_source_get_latency_within_thread(u->source_output->source) +
+ /* Add the latency internal to our source output on top */
+ pa_bytes_to_usec(pa_memblockq_get_length(u->source_output->thread_info.delay_memblockq), &u->source_output->source->sample_spec) +
+ /* and the buffering we do on the source */
+ pa_bytes_to_usec(u->blocksize, &u->source_output->source->sample_spec);
+
+ return 0;
+
+ }
+
+ return pa_source_process_msg(o, code, data, offset, chunk);
+}
+
+/* Called from sink I/O thread context */
+static int sink_process_msg_cb(pa_msgobject *o, int code, void *data, int64_t offset, pa_memchunk *chunk) {
+ struct userdata *u = PA_SINK(o)->userdata;
+
+ switch (code) {
+
+ case PA_SINK_MESSAGE_GET_LATENCY:
+
+ /* The sink is _put() before the sink input is, so let's
+ * make sure we don't access it in that time. Also, the
+ * sink input is first shut down, the sink second. */
+ if (!PA_SINK_IS_LINKED(u->sink->thread_info.state) ||
+ !PA_SINK_INPUT_IS_LINKED(u->sink_input->thread_info.state)) {
+ *((pa_usec_t*) data) = 0;
+ return 0;
+ }
+
+ *((pa_usec_t*) data) =
+
+ /* Get the latency of the master sink */
+ pa_sink_get_latency_within_thread(u->sink_input->sink) +
+
+ /* Add the latency internal to our sink input on top */
+ pa_bytes_to_usec(pa_memblockq_get_length(u->sink_input->thread_info.render_memblockq), &u->sink_input->sink->sample_spec);
+
+ return 0;
+ }
+
+ return pa_sink_process_msg(o, code, data, offset, chunk);
+}
+
+
+/* Called from main context */
+static int source_set_state_cb(pa_source *s, pa_source_state_t state) {
+ struct userdata *u;
+
+ pa_source_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SOURCE_IS_LINKED(state) ||
+ !PA_SOURCE_OUTPUT_IS_LINKED(pa_source_output_get_state(u->source_output)))
+ return 0;
+
+ pa_log_debug("Source state %d %d", state, u->active_mask);
+
+ if (state == PA_SOURCE_RUNNING) {
+ /* restart timer when both sink and source are active */
+ u->active_mask |= 1;
+ if (u->active_mask == 3)
+ pa_core_rttime_restart(u->core, u->time_event, pa_rtclock_now() + u->adjust_time);
+
+ pa_atomic_store(&u->request_resync, 1);
+ pa_source_output_cork(u->source_output, FALSE);
+ } else if (state == PA_SOURCE_SUSPENDED) {
+ u->active_mask &= ~1;
+ pa_source_output_cork(u->source_output, TRUE);
+ }
+ return 0;
+}
+
+/* Called from main context */
+static int sink_set_state_cb(pa_sink *s, pa_sink_state_t state) {
+ struct userdata *u;
+
+ pa_sink_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SINK_IS_LINKED(state) ||
+ !PA_SINK_INPUT_IS_LINKED(pa_sink_input_get_state(u->sink_input)))
+ return 0;
+
+ pa_log_debug("Sink state %d %d", state, u->active_mask);
+
+ if (state == PA_SINK_RUNNING) {
+ /* restart timer when both sink and source are active */
+ u->active_mask |= 2;
+ if (u->active_mask == 3)
+ pa_core_rttime_restart(u->core, u->time_event, pa_rtclock_now() + u->adjust_time);
+
+ pa_atomic_store(&u->request_resync, 1);
+ pa_sink_input_cork(u->sink_input, FALSE);
+ } else if (state == PA_SINK_SUSPENDED) {
+ u->active_mask &= ~2;
+ pa_sink_input_cork(u->sink_input, TRUE);
+ }
+ return 0;
+}
+
+/* Called from I/O thread context */
+static void source_update_requested_latency_cb(pa_source *s) {
+ struct userdata *u;
+
+ pa_source_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SOURCE_IS_LINKED(u->source->thread_info.state) ||
+ !PA_SOURCE_OUTPUT_IS_LINKED(u->source_output->thread_info.state))
+ return;
+
+ pa_log_debug("Source update requested latency");
+
+ /* Just hand this one over to the master source */
+ pa_source_output_set_requested_latency_within_thread(
+ u->source_output,
+ pa_source_get_requested_latency_within_thread(s));
+}
+
+/* Called from I/O thread context */
+static void sink_update_requested_latency_cb(pa_sink *s) {
+ struct userdata *u;
+
+ pa_sink_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SINK_IS_LINKED(u->sink->thread_info.state) ||
+ !PA_SINK_INPUT_IS_LINKED(u->sink_input->thread_info.state))
+ return;
+
+ pa_log_debug("Sink update requested latency");
+
+ /* Just hand this one over to the master sink */
+ pa_sink_input_set_requested_latency_within_thread(
+ u->sink_input,
+ pa_sink_get_requested_latency_within_thread(s));
+}
+
+/* Called from I/O thread context */
+static void sink_request_rewind_cb(pa_sink *s) {
+ struct userdata *u;
+
+ pa_sink_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SINK_IS_LINKED(u->sink->thread_info.state) ||
+ !PA_SINK_INPUT_IS_LINKED(u->sink_input->thread_info.state))
+ return;
+
+ pa_log_debug("Sink request rewind %lld", (long long) s->thread_info.rewind_nbytes);
+
+ /* Just hand this one over to the master sink */
+ pa_sink_input_request_rewind(u->sink_input,
+ s->thread_info.rewind_nbytes, TRUE, FALSE, FALSE);
+}
+
+/* Called from main context */
+static void source_set_volume_cb(pa_source *s) {
+ struct userdata *u;
+
+ pa_source_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SOURCE_IS_LINKED(pa_source_get_state(s)) ||
+ !PA_SOURCE_OUTPUT_IS_LINKED(pa_source_output_get_state(u->source_output)))
+ return;
+
+ pa_source_output_set_volume(u->source_output, &s->real_volume, s->save_volume, TRUE);
+}
+
+/* Called from main context */
+static void sink_set_volume_cb(pa_sink *s) {
+ struct userdata *u;
+
+ pa_sink_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SINK_IS_LINKED(pa_sink_get_state(s)) ||
+ !PA_SINK_INPUT_IS_LINKED(pa_sink_input_get_state(u->sink_input)))
+ return;
+
+ pa_sink_input_set_volume(u->sink_input, &s->real_volume, s->save_volume, TRUE);
+}
+
+static void source_get_volume_cb(pa_source *s) {
+ struct userdata *u;
+ pa_cvolume v;
+
+ pa_source_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SOURCE_IS_LINKED(pa_source_get_state(s)) ||
+ !PA_SOURCE_OUTPUT_IS_LINKED(pa_source_output_get_state(u->source_output)))
+ return;
+
+ pa_source_output_get_volume(u->source_output, &v, TRUE);
+
+ if (pa_cvolume_equal(&s->real_volume, &v))
+ /* no change */
+ return;
+
+ s->real_volume = v;
+ pa_source_set_soft_volume(s, NULL);
+}
+
+/* Called from main context */
+static void source_set_mute_cb(pa_source *s) {
+ struct userdata *u;
+
+ pa_source_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SOURCE_IS_LINKED(pa_source_get_state(s)) ||
+ !PA_SOURCE_OUTPUT_IS_LINKED(pa_source_output_get_state(u->source_output)))
+ return;
+
+ pa_source_output_set_mute(u->source_output, s->muted, s->save_muted);
+}
+
+/* Called from main context */
+static void sink_set_mute_cb(pa_sink *s) {
+ struct userdata *u;
+
+ pa_sink_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SINK_IS_LINKED(pa_sink_get_state(s)) ||
+ !PA_SINK_INPUT_IS_LINKED(pa_sink_input_get_state(u->sink_input)))
+ return;
+
+ pa_sink_input_set_mute(u->sink_input, s->muted, s->save_muted);
+}
+
+/* Called from main context */
+static void source_get_mute_cb(pa_source *s) {
+ struct userdata *u;
+
+ pa_source_assert_ref(s);
+ pa_assert_se(u = s->userdata);
+
+ if (!PA_SOURCE_IS_LINKED(pa_source_get_state(s)) ||
+ !PA_SOURCE_OUTPUT_IS_LINKED(pa_source_output_get_state(u->source_output)))
+ return;
+
+ pa_source_output_get_mute(u->source_output);
+}
+
+/* must be called from the input thread context */
+static void apply_diff_time(struct userdata *u, int64_t diff_time) {
+ int64_t diff;
+
+ if (diff_time < 0) {
+ diff = pa_usec_to_bytes(-diff_time, &u->source_output->sample_spec);
+
+ if (diff > 0) {
+ /* add some extra safety samples to compensate for jitter in the
+ * timings */
+ diff += 10 * pa_frame_size (&u->source_output->sample_spec);
+
+ pa_log("Playback after capture (%lld), drop sink %lld", (long long) diff_time, (long long) diff);
+
+ u->sink_skip = diff;
+ u->source_skip = 0;
+ }
+ } else if (diff_time > 0) {
+ diff = pa_usec_to_bytes(diff_time, &u->source_output->sample_spec);
+
+ if (diff > 0) {
+ pa_log("playback too far ahead (%lld), drop source %lld", (long long) diff_time, (long long) diff);
+
+ u->source_skip = diff;
+ u->sink_skip = 0;
+ }
+ }
+}
+
+/* must be called from the input thread */
+static void do_resync(struct userdata *u) {
+ int64_t diff_time;
+ struct snapshot latency_snapshot;
+
+ pa_log("Doing resync");
+
+ /* update our snapshot */
+ source_output_snapshot_within_thread(u, &latency_snapshot);
+ pa_asyncmsgq_send(u->sink_input->sink->asyncmsgq, PA_MSGOBJECT(u->sink_input), SINK_INPUT_MESSAGE_LATENCY_SNAPSHOT, &latency_snapshot, 0, NULL);
+
+ /* calculate drift between capture and playback */
+ diff_time = calc_diff(u, &latency_snapshot);
+
+ /* and adjust for the drift */
+ apply_diff_time(u, diff_time);
+}
+
+/* Called from input thread context */
+static void source_output_push_cb(pa_source_output *o, const pa_memchunk *chunk) {
+ struct userdata *u;
+ size_t rlen, plen;
+
+ pa_source_output_assert_ref(o);
+ pa_source_output_assert_io_context(o);
+ pa_assert_se(u = o->userdata);
+
+ if (!PA_SOURCE_OUTPUT_IS_LINKED(pa_source_output_get_state(u->source_output))) {
+ pa_log("push when no link?");
+ return;
+ }
+
+ /* handle queued messages */
+ u->in_push = TRUE;
+ while (pa_asyncmsgq_process_one(u->asyncmsgq) > 0)
+ ;
+ u->in_push = FALSE;
+
+ if (pa_atomic_cmpxchg (&u->request_resync, 1, 0)) {
+ do_resync(u);
+ }
+
+ pa_memblockq_push_align(u->source_memblockq, chunk);
+
+ rlen = pa_memblockq_get_length(u->source_memblockq);
+ plen = pa_memblockq_get_length(u->sink_memblockq);
+
+ while (rlen >= u->blocksize) {
+ pa_memchunk rchunk, pchunk;
+
+ /* take fixed block from recorded samples */
+ pa_memblockq_peek_fixed_size(u->source_memblockq, u->blocksize, &rchunk);
+
+ if (plen > u->blocksize && u->source_skip == 0) {
+ uint8_t *rdata, *pdata, *cdata;
+ pa_memchunk cchunk;
+
+ if (u->sink_skip) {
+ size_t to_skip;
+
+ if (u->sink_skip > plen)
+ to_skip = plen;
+ else
+ to_skip = u->sink_skip;
+
+ pa_memblockq_drop(u->sink_memblockq, to_skip);
+ plen -= to_skip;
+
+ u->sink_skip -= to_skip;
+ }
+
+ if (plen > u->blocksize && u->sink_skip == 0) {
+ /* take fixed block from played samples */
+ pa_memblockq_peek_fixed_size(u->sink_memblockq, u->blocksize, &pchunk);
+
+ rdata = pa_memblock_acquire(rchunk.memblock);
+ rdata += rchunk.index;
+ pdata = pa_memblock_acquire(pchunk.memblock);
+ pdata += pchunk.index;
+
+ cchunk.index = 0;
+ cchunk.length = u->blocksize;
+ cchunk.memblock = pa_memblock_new(u->source->core->mempool, cchunk.length);
+ cdata = pa_memblock_acquire(cchunk.memblock);
+
+ if (u->save_aec) {
+ if (u->captured_file)
+ fwrite(rdata, 1, u->blocksize, u->captured_file);
+ if (u->played_file)
+ fwrite(pdata, 1, u->blocksize, u->played_file);
+ }
+
+ /* perform echo cancelation */
+ u->ec->run(u->ec, rdata, pdata, cdata);
+
+ /* preprecessor is run after AEC. This is not a mistake! */
+ if (u->ec->pp_state)
+ speex_preprocess_run(u->ec->pp_state, (spx_int16_t *) cdata);
+
+ if (u->save_aec) {
+ if (u->canceled_file)
+ fwrite(cdata, 1, u->blocksize, u->canceled_file);
+ }
+
+ pa_memblock_release(cchunk.memblock);
+ pa_memblock_release(pchunk.memblock);
+ pa_memblock_release(rchunk.memblock);
+
+ /* drop consumed sink samples */
+ pa_memblockq_drop(u->sink_memblockq, u->blocksize);
+ pa_memblock_unref(pchunk.memblock);
+
+ pa_memblock_unref(rchunk.memblock);
+ /* the filtered samples now become the samples from our
+ * source */
+ rchunk = cchunk;
+
+ plen -= u->blocksize;
+ }
+ }
+
+ /* forward the (echo-canceled) data to the virtual source */
+ pa_source_post(u->source, &rchunk);
+ pa_memblock_unref(rchunk.memblock);
+
+ pa_memblockq_drop(u->source_memblockq, u->blocksize);
+ rlen -= u->blocksize;
+
+ if (u->source_skip) {
+ if (u->source_skip > u->blocksize) {
+ u->source_skip -= u->blocksize;
+ }
+ else {
+ u->sink_skip += (u->blocksize - u->source_skip);
+ u->source_skip = 0;
+ }
+ }
+ }
+}
+
+/* Called from I/O thread context */
+static int sink_input_pop_cb(pa_sink_input *i, size_t nbytes, pa_memchunk *chunk) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert(chunk);
+ pa_assert_se(u = i->userdata);
+
+ if (u->sink->thread_info.rewind_requested)
+ pa_sink_process_rewind(u->sink, 0);
+
+ pa_sink_render_full(u->sink, nbytes, chunk);
+
+ if (i->thread_info.underrun_for > 0) {
+ pa_log_debug("Handling end of underrun.");
+ pa_atomic_store(&u->request_resync, 1);
+ }
+
+ /* let source thread handle the chunk. pass the sample count as well so that
+ * the source IO thread can update the right variables. */
+ pa_asyncmsgq_post(u->asyncmsgq, PA_MSGOBJECT(u->source_output), SOURCE_OUTPUT_MESSAGE_POST,
+ NULL, 0, chunk, NULL);
+ u->send_counter += chunk->length;
+
+ return 0;
+}
+
+/* Called from input thread context */
+static void source_output_process_rewind_cb(pa_source_output *o, size_t nbytes) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_source_output_assert_io_context(o);
+ pa_assert_se(u = o->userdata);
+
+ pa_source_process_rewind(u->source, nbytes);
+
+ /* go back on read side, we need to use older sink data for this */
+ pa_memblockq_rewind(u->sink_memblockq, nbytes);
+
+ /* manipulate write index */
+ pa_memblockq_seek(u->source_memblockq, -nbytes, PA_SEEK_RELATIVE, TRUE);
+
+ pa_log_debug("Source rewind (%lld) %lld", (long long) nbytes,
+ (long long) pa_memblockq_get_length (u->source_memblockq));
+}
+
+/* Called from I/O thread context */
+static void sink_input_process_rewind_cb(pa_sink_input *i, size_t nbytes) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_log_debug("Sink process rewind %lld", (long long) nbytes);
+
+ pa_sink_process_rewind(u->sink, nbytes);
+
+ pa_asyncmsgq_post(u->asyncmsgq, PA_MSGOBJECT(u->source_output), SOURCE_OUTPUT_MESSAGE_REWIND, NULL, (int64_t) nbytes, NULL, NULL);
+ u->send_counter -= nbytes;
+}
+
+static void source_output_snapshot_within_thread(struct userdata *u, struct snapshot *snapshot) {
+ size_t delay, rlen, plen;
+ pa_usec_t now, latency;
+
+ now = pa_rtclock_now();
+ latency = pa_source_get_latency_within_thread(u->source_output->source);
+ delay = pa_memblockq_get_length(u->source_output->thread_info.delay_memblockq);
+
+ delay = (u->source_output->thread_info.resampler ? pa_resampler_request(u->source_output->thread_info.resampler, delay) : delay);
+ rlen = pa_memblockq_get_length(u->source_memblockq);
+ plen = pa_memblockq_get_length(u->sink_memblockq);
+
+ snapshot->source_now = now;
+ snapshot->source_latency = latency;
+ snapshot->source_delay = delay;
+ snapshot->recv_counter = u->recv_counter;
+ snapshot->rlen = rlen + u->sink_skip;
+ snapshot->plen = plen + u->source_skip;
+}
+
+
+/* Called from output thread context */
+static int source_output_process_msg_cb(pa_msgobject *obj, int code, void *data, int64_t offset, pa_memchunk *chunk) {
+ struct userdata *u = PA_SOURCE_OUTPUT(obj)->userdata;
+
+ switch (code) {
+
+ case SOURCE_OUTPUT_MESSAGE_POST:
+
+ pa_source_output_assert_io_context(u->source_output);
+
+ if (PA_SOURCE_IS_OPENED(u->source_output->source->thread_info.state))
+ pa_memblockq_push_align(u->sink_memblockq, chunk);
+ else
+ pa_memblockq_flush_write(u->sink_memblockq, TRUE);
+
+ u->recv_counter += (int64_t) chunk->length;
+
+ return 0;
+
+ case SOURCE_OUTPUT_MESSAGE_REWIND:
+ pa_source_output_assert_io_context(u->source_output);
+
+ /* manipulate write index, never go past what we have */
+ if (PA_SOURCE_IS_OPENED(u->source_output->source->thread_info.state))
+ pa_memblockq_seek(u->sink_memblockq, -offset, PA_SEEK_RELATIVE, TRUE);
+ else
+ pa_memblockq_flush_write(u->sink_memblockq, TRUE);
+
+ pa_log_debug("Sink rewind (%lld)", (long long) offset);
+
+ u->recv_counter -= offset;
+
+ return 0;
+
+ case SOURCE_OUTPUT_MESSAGE_LATENCY_SNAPSHOT: {
+ struct snapshot *snapshot = (struct snapshot *) data;
+
+ source_output_snapshot_within_thread(u, snapshot);
+ return 0;
+ }
+
+ case SOURCE_OUTPUT_MESSAGE_APPLY_DIFF_TIME:
+ apply_diff_time(u, offset);
+ return 0;
+
+ }
+
+ return pa_source_output_process_msg(obj, code, data, offset, chunk);
+}
+
+static int sink_input_process_msg_cb(pa_msgobject *obj, int code, void *data, int64_t offset, pa_memchunk *chunk) {
+ struct userdata *u = PA_SINK_INPUT(obj)->userdata;
+
+ switch (code) {
+
+ case SINK_INPUT_MESSAGE_LATENCY_SNAPSHOT: {
+ size_t delay;
+ pa_usec_t now, latency;
+ struct snapshot *snapshot = (struct snapshot *) data;
+
+ pa_sink_input_assert_io_context(u->sink_input);
+
+ now = pa_rtclock_now();
+ latency = pa_sink_get_latency_within_thread(u->sink_input->sink);
+ delay = pa_memblockq_get_length(u->sink_input->thread_info.render_memblockq);
+
+ delay = (u->sink_input->thread_info.resampler ? pa_resampler_request(u->sink_input->thread_info.resampler, delay) : delay);
+
+ snapshot->sink_now = now;
+ snapshot->sink_latency = latency;
+ snapshot->sink_delay = delay;
+ snapshot->send_counter = u->send_counter;
+ return 0;
+ }
+ }
+
+ return pa_sink_input_process_msg(obj, code, data, offset, chunk);
+}
+
+/* Called from I/O thread context */
+static void sink_input_update_max_rewind_cb(pa_sink_input *i, size_t nbytes) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_log_debug("Sink input update max rewind %lld", (long long) nbytes);
+
+ pa_memblockq_set_maxrewind(u->sink_memblockq, nbytes);
+ pa_sink_set_max_rewind_within_thread(u->sink, nbytes);
+}
+
+/* Called from I/O thread context */
+static void source_output_update_max_rewind_cb(pa_source_output *o, size_t nbytes) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_assert_se(u = o->userdata);
+
+ pa_log_debug("Source output update max rewind %lld", (long long) nbytes);
+
+ pa_source_set_max_rewind_within_thread(u->source, nbytes);
+}
+
+/* Called from I/O thread context */
+static void sink_input_update_max_request_cb(pa_sink_input *i, size_t nbytes) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_log_debug("Sink input update max request %lld", (long long) nbytes);
+
+ pa_sink_set_max_request_within_thread(u->sink, nbytes);
+}
+
+/* Called from I/O thread context */
+static void sink_input_update_sink_requested_latency_cb(pa_sink_input *i) {
+ struct userdata *u;
+ pa_usec_t latency;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ latency = pa_sink_get_requested_latency_within_thread(i->sink);
+
+ pa_log_debug("Sink input update requested latency %lld", (long long) latency);
+}
+
+/* Called from I/O thread context */
+static void source_output_update_source_requested_latency_cb(pa_source_output *o) {
+ struct userdata *u;
+ pa_usec_t latency;
+
+ pa_source_output_assert_ref(o);
+ pa_assert_se(u = o->userdata);
+
+ latency = pa_source_get_requested_latency_within_thread(o->source);
+
+ pa_log_debug("source output update requested latency %lld", (long long) latency);
+}
+
+/* Called from I/O thread context */
+static void sink_input_update_sink_latency_range_cb(pa_sink_input *i) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_log_debug("Sink input update latency range %lld %lld",
+ (long long) i->sink->thread_info.min_latency,
+ (long long) i->sink->thread_info.max_latency);
+
+ pa_sink_set_latency_range_within_thread(u->sink, i->sink->thread_info.min_latency, i->sink->thread_info.max_latency);
+}
+
+/* Called from I/O thread context */
+static void source_output_update_source_latency_range_cb(pa_source_output *o) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_assert_se(u = o->userdata);
+
+ pa_log_debug("Source output update latency range %lld %lld",
+ (long long) o->source->thread_info.min_latency,
+ (long long) o->source->thread_info.max_latency);
+
+ pa_source_set_latency_range_within_thread(u->source, o->source->thread_info.min_latency, o->source->thread_info.max_latency);
+}
+
+/* Called from I/O thread context */
+static void sink_input_update_sink_fixed_latency_cb(pa_sink_input *i) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_log_debug("Sink input update fixed latency %lld",
+ (long long) i->sink->thread_info.fixed_latency);
+
+ pa_sink_set_fixed_latency_within_thread(u->sink, i->sink->thread_info.fixed_latency);
+}
+
+/* Called from I/O thread context */
+static void source_output_update_source_fixed_latency_cb(pa_source_output *o) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_assert_se(u = o->userdata);
+
+ pa_log_debug("Source output update fixed latency %lld",
+ (long long) o->source->thread_info.fixed_latency);
+
+ pa_source_set_fixed_latency_within_thread(u->source, o->source->thread_info.fixed_latency);
+}
+
+/* Called from output thread context */
+static void source_output_attach_cb(pa_source_output *o) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_source_output_assert_io_context(o);
+ pa_assert_se(u = o->userdata);
+
+ pa_source_set_rtpoll(u->source, o->source->thread_info.rtpoll);
+ pa_source_set_latency_range_within_thread(u->source, o->source->thread_info.min_latency, o->source->thread_info.max_latency);
+ pa_source_set_fixed_latency_within_thread(u->source, o->source->thread_info.fixed_latency);
+ pa_source_set_max_rewind_within_thread(u->source, pa_source_output_get_max_rewind(o));
+
+ pa_log_debug("Source output %p attach", o);
+
+ pa_source_attach_within_thread(u->source);
+
+ u->rtpoll_item_read = pa_rtpoll_item_new_asyncmsgq_read(
+ o->source->thread_info.rtpoll,
+ PA_RTPOLL_LATE,
+ u->asyncmsgq);
+}
+
+/* Called from I/O thread context */
+static void sink_input_attach_cb(pa_sink_input *i) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_sink_set_rtpoll(u->sink, i->sink->thread_info.rtpoll);
+ pa_sink_set_latency_range_within_thread(u->sink, i->sink->thread_info.min_latency, i->sink->thread_info.max_latency);
+
+ /* (8.1) IF YOU NEED A FIXED BLOCK SIZE ADD THE LATENCY FOR ONE
+ * BLOCK MINUS ONE SAMPLE HERE. SEE (7) */
+ pa_sink_set_fixed_latency_within_thread(u->sink, i->sink->thread_info.fixed_latency);
+
+ /* (8.2) IF YOU NEED A FIXED BLOCK SIZE ROUND
+ * pa_sink_input_get_max_request(i) UP TO MULTIPLES OF IT
+ * HERE. SEE (6) */
+ pa_sink_set_max_request_within_thread(u->sink, pa_sink_input_get_max_request(i));
+ pa_sink_set_max_rewind_within_thread(u->sink, pa_sink_input_get_max_rewind(i));
+
+ pa_log_debug("Sink input %p attach", i);
+
+ u->rtpoll_item_write = pa_rtpoll_item_new_asyncmsgq_write(
+ i->sink->thread_info.rtpoll,
+ PA_RTPOLL_LATE,
+ u->asyncmsgq);
+
+ pa_sink_attach_within_thread(u->sink);
+}
+
+
+/* Called from output thread context */
+static void source_output_detach_cb(pa_source_output *o) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_source_output_assert_io_context(o);
+ pa_assert_se(u = o->userdata);
+
+ pa_source_detach_within_thread(u->source);
+ pa_source_set_rtpoll(u->source, NULL);
+
+ pa_log_debug("Source output %p detach", o);
+
+ if (u->rtpoll_item_read) {
+ pa_rtpoll_item_free(u->rtpoll_item_read);
+ u->rtpoll_item_read = NULL;
+ }
+}
+
+/* Called from I/O thread context */
+static void sink_input_detach_cb(pa_sink_input *i) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_sink_detach_within_thread(u->sink);
+
+ pa_sink_set_rtpoll(u->sink, NULL);
+
+ pa_log_debug("Sink input %p detach", i);
+
+ if (u->rtpoll_item_write) {
+ pa_rtpoll_item_free(u->rtpoll_item_write);
+ u->rtpoll_item_write = NULL;
+ }
+}
+
+/* Called from output thread context */
+static void source_output_state_change_cb(pa_source_output *o, pa_source_output_state_t state) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_source_output_assert_io_context(o);
+ pa_assert_se(u = o->userdata);
+
+ pa_log_debug("Source output %p state %d", o, state);
+}
+
+/* Called from IO thread context */
+static void sink_input_state_change_cb(pa_sink_input *i, pa_sink_input_state_t state) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_log_debug("Sink input %p state %d", i, state);
+
+ /* If we are added for the first time, ask for a rewinding so that
+ * we are heard right-away. */
+ if (PA_SINK_INPUT_IS_LINKED(state) &&
+ i->thread_info.state == PA_SINK_INPUT_INIT) {
+ pa_log_debug("Requesting rewind due to state change.");
+ pa_sink_input_request_rewind(i, 0, FALSE, TRUE, TRUE);
+ }
+}
+
+/* Called from main thread */
+static void source_output_kill_cb(pa_source_output *o) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_assert_ctl_context();
+ pa_assert_se(u = o->userdata);
+
+ /* The order here matters! We first kill the source output, followed
+ * by the source. That means the source callbacks must be protected
+ * against an unconnected source output! */
+ pa_source_output_unlink(u->source_output);
+ pa_source_unlink(u->source);
+
+ pa_source_output_unref(u->source_output);
+ u->source_output = NULL;
+
+ pa_source_unref(u->source);
+ u->source = NULL;
+
+ pa_log_debug("Source output kill %p", o);
+
+ pa_module_unload_request(u->module, TRUE);
+}
+
+/* Called from main context */
+static void sink_input_kill_cb(pa_sink_input *i) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ /* The order here matters! We first kill the sink input, followed
+ * by the sink. That means the sink callbacks must be protected
+ * against an unconnected sink input! */
+ pa_sink_input_unlink(u->sink_input);
+ pa_sink_unlink(u->sink);
+
+ pa_sink_input_unref(u->sink_input);
+ u->sink_input = NULL;
+
+ pa_sink_unref(u->sink);
+ u->sink = NULL;
+
+ pa_log_debug("Sink input kill %p", i);
+
+ pa_module_unload_request(u->module, TRUE);
+}
+
+/* Called from main thread */
+static pa_bool_t source_output_may_move_to_cb(pa_source_output *o, pa_source *dest) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_assert_ctl_context();
+ pa_assert_se(u = o->userdata);
+
+ return (u->source != dest) && (u->sink != dest->monitor_of);
+}
+
+/* Called from main context */
+static pa_bool_t sink_input_may_move_to_cb(pa_sink_input *i, pa_sink *dest) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ return u->sink != dest;
+}
+
+/* Called from main thread */
+static void source_output_moving_cb(pa_source_output *o, pa_source *dest) {
+ struct userdata *u;
+
+ pa_source_output_assert_ref(o);
+ pa_assert_ctl_context();
+ pa_assert_se(u = o->userdata);
+
+ if (dest) {
+ pa_source_set_asyncmsgq(u->source, dest->asyncmsgq);
+ pa_source_update_flags(u->source, PA_SOURCE_LATENCY|PA_SOURCE_DYNAMIC_LATENCY, dest->flags);
+ } else
+ pa_source_set_asyncmsgq(u->source, NULL);
+
+ if (u->source_auto_desc && dest) {
+ const char *z;
+ pa_proplist *pl;
+
+ pl = pa_proplist_new();
+ z = pa_proplist_gets(dest->proplist, PA_PROP_DEVICE_DESCRIPTION);
+ pa_proplist_setf(pl, PA_PROP_DEVICE_DESCRIPTION, "Echo-Cancel Source %s on %s",
+ pa_proplist_gets(u->source->proplist, "device.echo-cancel.name"), z ? z : dest->name);
+
+ pa_source_update_proplist(u->source, PA_UPDATE_REPLACE, pl);
+ pa_proplist_free(pl);
+ }
+}
+
+/* Called from main context */
+static void sink_input_moving_cb(pa_sink_input *i, pa_sink *dest) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ if (dest) {
+ pa_sink_set_asyncmsgq(u->sink, dest->asyncmsgq);
+ pa_sink_update_flags(u->sink, PA_SINK_LATENCY|PA_SINK_DYNAMIC_LATENCY, dest->flags);
+ } else
+ pa_sink_set_asyncmsgq(u->sink, NULL);
+
+ if (u->sink_auto_desc && dest) {
+ const char *z;
+ pa_proplist *pl;
+
+ pl = pa_proplist_new();
+ z = pa_proplist_gets(dest->proplist, PA_PROP_DEVICE_DESCRIPTION);
+ pa_proplist_setf(pl, PA_PROP_DEVICE_DESCRIPTION, "Echo-Cancel Sink %s on %s",
+ pa_proplist_gets(u->sink->proplist, "device.echo-cancel.name"), z ? z : dest->name);
+
+ pa_sink_update_proplist(u->sink, PA_UPDATE_REPLACE, pl);
+ pa_proplist_free(pl);
+ }
+}
+
+/* Called from main context */
+static void sink_input_volume_changed_cb(pa_sink_input *i) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_sink_volume_changed(u->sink, &i->volume);
+}
+
+/* Called from main context */
+static void sink_input_mute_changed_cb(pa_sink_input *i) {
+ struct userdata *u;
+
+ pa_sink_input_assert_ref(i);
+ pa_assert_se(u = i->userdata);
+
+ pa_sink_mute_changed(u->sink, i->muted);
+}
+
+static pa_echo_canceller_method_t get_ec_method_from_string(const char *method) {
+ if (strcmp(method, "speex") == 0)
+ return PA_ECHO_CANCELLER_SPEEX;
+ else if (strcmp(method, "adrian") == 0)
+ return PA_ECHO_CANCELLER_ADRIAN;
+ else
+ return PA_ECHO_CANCELLER_INVALID;
+}
+
+int pa__init(pa_module*m) {
+ struct userdata *u;
+ pa_sample_spec source_ss, sink_ss;
+ pa_channel_map source_map, sink_map;
+ pa_modargs *ma;
+ pa_source *source_master=NULL;
+ pa_sink *sink_master=NULL;
+ pa_source_output_new_data source_output_data;
+ pa_sink_input_new_data sink_input_data;
+ pa_source_new_data source_data;
+ pa_sink_new_data sink_data;
+ pa_memchunk silence;
+ pa_echo_canceller_method_t ec_method;
+ uint32_t adjust_time_sec;
+
+ pa_assert(m);
+
+ if (!(ma = pa_modargs_new(m->argument, valid_modargs))) {
+ pa_log("Failed to parse module arguments.");
+ goto fail;
+ }
+
+ if (!(source_master = pa_namereg_get(m->core, pa_modargs_get_value(ma, "source_master", NULL), PA_NAMEREG_SOURCE))) {
+ pa_log("Master source not found");
+ goto fail;
+ }
+ pa_assert(source_master);
+
+ if (!(sink_master = pa_namereg_get(m->core, pa_modargs_get_value(ma, "sink_master", NULL), PA_NAMEREG_SINK))) {
+ pa_log("Master sink not found");
+ goto fail;
+ }
+ pa_assert(sink_master);
+
+ source_ss = source_master->sample_spec;
+ source_map = source_master->channel_map;
+ if (pa_modargs_get_sample_spec_and_channel_map(ma, &source_ss, &source_map, PA_CHANNEL_MAP_DEFAULT) < 0) {
+ pa_log("Invalid sample format specification or channel map");
+ goto fail;
+ }
+
+ sink_ss = sink_master->sample_spec;
+ sink_map = sink_master->channel_map;
+
+ u = pa_xnew0(struct userdata, 1);
+ if (!u) {
+ pa_log("Failed to alloc userdata");
+ goto fail;
+ }
+ u->core = m->core;
+ u->module = m;
+ m->userdata = u;
+
+ u->ec = pa_xnew0(pa_echo_canceller, 1);
+ if (!u->ec) {
+ pa_log("Failed to alloc echo canceller");
+ goto fail;
+ }
+
+ if ((ec_method = get_ec_method_from_string(pa_modargs_get_value(ma, "aec_method", DEFAULT_ECHO_CANCELLER))) < 0) {
+ pa_log("Invalid echo canceller implementation");
+ goto fail;
+ }
+
+ u->ec->init = ec_table[ec_method].init;
+ u->ec->run = ec_table[ec_method].run;
+ u->ec->done = ec_table[ec_method].done;
+
+ adjust_time_sec = DEFAULT_ADJUST_TIME_USEC / PA_USEC_PER_SEC;
+ if (pa_modargs_get_value_u32(ma, "adjust_time", &adjust_time_sec) < 0) {
+ pa_log("Failed to parse adjust_time value");
+ goto fail;
+ }
+
+ if (adjust_time_sec != DEFAULT_ADJUST_TIME_USEC / PA_USEC_PER_SEC)
+ u->adjust_time = adjust_time_sec * PA_USEC_PER_SEC;
+ else
+ u->adjust_time = DEFAULT_ADJUST_TIME_USEC;
+
+ u->ec->agc = DEFAULT_AGC_ENABLED;
+ if (pa_modargs_get_value_boolean(ma, "agc", &u->ec->agc) < 0) {
+ pa_log("Failed to parse agc value");
+ goto fail;
+ }
+
+ u->ec->denoise = DEFAULT_DENOISE_ENABLED;
+ if (pa_modargs_get_value_boolean(ma, "denoise", &u->ec->denoise) < 0) {
+ pa_log("Failed to parse denoise value");
+ goto fail;
+ }
+
+ u->ec->echo_suppress = DEFAULT_ECHO_SUPPRESS_ENABLED;
+ if (pa_modargs_get_value_boolean(ma, "echo_suppress", &u->ec->echo_suppress) < 0) {
+ pa_log("Failed to parse echo_suppress value");
+ goto fail;
+ }
+ if (u->ec->echo_suppress && ec_method != PA_ECHO_CANCELLER_SPEEX) {
+ pa_log("Echo suppression is only useful with the speex canceller");
+ goto fail;
+ }
+
+ u->ec->echo_suppress_attenuation = DEFAULT_ECHO_SUPPRESS_ATTENUATION;
+ if (pa_modargs_get_value_s32(ma, "echo_suppress_attenuation", &u->ec->echo_suppress_attenuation) < 0) {
+ pa_log("Failed to parse echo_suppress_attenuation value");
+ goto fail;
+ }
+ if (u->ec->echo_suppress_attenuation > 0) {
+ pa_log("echo_suppress_attenuation should be a negative dB value");
+ goto fail;
+ }
+
+ u->ec->echo_suppress_attenuation_active = DEFAULT_ECHO_SUPPRESS_ATTENUATION;
+ if (pa_modargs_get_value_s32(ma, "echo_suppress_attenuation_active", &u->ec->echo_suppress_attenuation_active) < 0) {
+ pa_log("Failed to parse echo_supress_attenuation_active value");
+ goto fail;
+ }
+ if (u->ec->echo_suppress_attenuation_active > 0) {
+ pa_log("echo_suppress_attenuation_active should be a negative dB value");
+ goto fail;
+ }
+
+ u->save_aec = DEFAULT_SAVE_AEC;
+ if (pa_modargs_get_value_u32(ma, "save_aec", &u->save_aec) < 0) {
+ pa_log("Failed to parse save_aec value");
+ goto fail;
+ }
+
+ u->autoloaded = DEFAULT_AUTOLOADED;
+ if (pa_modargs_get_value_boolean(ma, "autoloaded", &u->autoloaded) < 0) {
+ pa_log("Failed to parse autoloaded value");
+ goto fail;
+ }
+
+ u->asyncmsgq = pa_asyncmsgq_new(0);
+ u->need_realign = TRUE;
+ if (u->ec->init) {
+ if (!u->ec->init(u->core, u->ec, &source_ss, &source_map, &sink_ss, &sink_map, &u->blocksize, pa_modargs_get_value(ma, "aec_args", NULL))) {
+ pa_log("Failed to init AEC engine");
+ goto fail;
+ }
+ }
+
+ if (u->ec->agc || u->ec->denoise || u->ec->echo_suppress) {
+ spx_int32_t tmp;
+
+ if (source_ss.channels != 1) {
+ pa_log("AGC, denoising and echo suppression only work with channels=1");
+ goto fail;
+ }
+
+ u->ec->pp_state = speex_preprocess_state_init(u->blocksize / pa_frame_size(&source_ss), source_ss.rate);
+
+ tmp = u->ec->agc;
+ speex_preprocess_ctl(u->ec->pp_state, SPEEX_PREPROCESS_SET_AGC, &tmp);
+ tmp = u->ec->denoise;
+ speex_preprocess_ctl(u->ec->pp_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp);
+ if (u->ec->echo_suppress) {
+ if (u->ec->echo_suppress_attenuation)
+ speex_preprocess_ctl(u->ec->pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS, &u->ec->echo_suppress_attenuation);
+ if (u->ec->echo_suppress_attenuation_active) {
+ speex_preprocess_ctl(u->ec->pp_state, SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE,
+ &u->ec->echo_suppress_attenuation_active);
+ }
+ speex_preprocess_ctl(u->ec->pp_state, SPEEX_PREPROCESS_SET_ECHO_STATE, u->ec->params.priv.speex.state);
+ }
+ }
+
+ /* Create source */
+ pa_source_new_data_init(&source_data);
+ source_data.driver = __FILE__;
+ source_data.module = m;
+ if (!(source_data.name = pa_xstrdup(pa_modargs_get_value(ma, "source_name", NULL))))
+ source_data.name = pa_sprintf_malloc("%s.echo-cancel", source_master->name);
+ pa_source_new_data_set_sample_spec(&source_data, &source_ss);
+ pa_source_new_data_set_channel_map(&source_data, &source_map);
+ pa_proplist_sets(source_data.proplist, PA_PROP_DEVICE_MASTER_DEVICE, source_master->name);
+ pa_proplist_sets(source_data.proplist, PA_PROP_DEVICE_CLASS, "filter");
+ if (!u->autoloaded)
+ pa_proplist_sets(source_data.proplist, PA_PROP_DEVICE_INTENDED_ROLES, "phone");
+ pa_proplist_sets(source_data.proplist, "device.echo-cancel.name", source_data.name);
+
+ if (pa_modargs_get_proplist(ma, "source_properties", source_data.proplist, PA_UPDATE_REPLACE) < 0) {
+ pa_log("Invalid properties");
+ pa_source_new_data_done(&source_data);
+ goto fail;
+ }
+
+ if ((u->source_auto_desc = !pa_proplist_contains(source_data.proplist, PA_PROP_DEVICE_DESCRIPTION))) {
+ const char *z;
+
+ z = pa_proplist_gets(source_master->proplist, PA_PROP_DEVICE_DESCRIPTION);
+ pa_proplist_setf(source_data.proplist, PA_PROP_DEVICE_DESCRIPTION, "Echo-Cancel Source %s on %s", source_data.name, z ? z : source_master->name);
+ }
+
+ u->source = pa_source_new(m->core, &source_data,
+ PA_SOURCE_HW_MUTE_CTRL|PA_SOURCE_HW_VOLUME_CTRL|PA_SOURCE_DECIBEL_VOLUME|
+ (source_master->flags & (PA_SOURCE_LATENCY|PA_SOURCE_DYNAMIC_LATENCY)));
+ pa_source_new_data_done(&source_data);
+
+ if (!u->source) {
+ pa_log("Failed to create source.");
+ goto fail;
+ }
+
+ u->source->parent.process_msg = source_process_msg_cb;
+ u->source->set_state = source_set_state_cb;
+ u->source->update_requested_latency = source_update_requested_latency_cb;
+ u->source->set_volume = source_set_volume_cb;
+ u->source->set_mute = source_set_mute_cb;
+ u->source->get_volume = source_get_volume_cb;
+ u->source->get_mute = source_get_mute_cb;
+ u->source->userdata = u;
+
+ pa_source_set_asyncmsgq(u->source, source_master->asyncmsgq);
+
+ /* Create sink */
+ pa_sink_new_data_init(&sink_data);
+ sink_data.driver = __FILE__;
+ sink_data.module = m;
+ if (!(sink_data.name = pa_xstrdup(pa_modargs_get_value(ma, "sink_name", NULL))))
+ sink_data.name = pa_sprintf_malloc("%s.echo-cancel", sink_master->name);
+ pa_sink_new_data_set_sample_spec(&sink_data, &sink_ss);
+ pa_sink_new_data_set_channel_map(&sink_data, &sink_map);
+ pa_proplist_sets(sink_data.proplist, PA_PROP_DEVICE_MASTER_DEVICE, sink_master->name);
+ pa_proplist_sets(sink_data.proplist, PA_PROP_DEVICE_CLASS, "filter");
+ if (!u->autoloaded)
+ pa_proplist_sets(sink_data.proplist, PA_PROP_DEVICE_INTENDED_ROLES, "phone");
+ pa_proplist_sets(sink_data.proplist, "device.echo-cancel.name", sink_data.name);
+
+ if (pa_modargs_get_proplist(ma, "sink_properties", sink_data.proplist, PA_UPDATE_REPLACE) < 0) {
+ pa_log("Invalid properties");
+ pa_sink_new_data_done(&sink_data);
+ goto fail;
+ }
+
+ if ((u->sink_auto_desc = !pa_proplist_contains(sink_data.proplist, PA_PROP_DEVICE_DESCRIPTION))) {
+ const char *z;
+
+ z = pa_proplist_gets(sink_master->proplist, PA_PROP_DEVICE_DESCRIPTION);
+ pa_proplist_setf(sink_data.proplist, PA_PROP_DEVICE_DESCRIPTION, "Echo-Cancel Sink %s on %s", sink_data.name, z ? z : sink_master->name);
+ }
+
+ u->sink = pa_sink_new(m->core, &sink_data,
+ PA_SINK_HW_MUTE_CTRL|PA_SINK_HW_VOLUME_CTRL|PA_SINK_DECIBEL_VOLUME|
+ (sink_master->flags & (PA_SINK_LATENCY|PA_SINK_DYNAMIC_LATENCY)));
+ pa_sink_new_data_done(&sink_data);
+
+ if (!u->sink) {
+ pa_log("Failed to create sink.");
+ goto fail;
+ }
+
+ u->sink->parent.process_msg = sink_process_msg_cb;
+ u->sink->set_state = sink_set_state_cb;
+ u->sink->update_requested_latency = sink_update_requested_latency_cb;
+ u->sink->request_rewind = sink_request_rewind_cb;
+ u->sink->set_volume = sink_set_volume_cb;
+ u->sink->set_mute = sink_set_mute_cb;
+ u->sink->userdata = u;
+
+ pa_sink_set_asyncmsgq(u->sink, sink_master->asyncmsgq);
+
+ /* Create source output */
+ pa_source_output_new_data_init(&source_output_data);
+ source_output_data.driver = __FILE__;
+ source_output_data.module = m;
+ pa_source_output_new_data_set_source(&source_output_data, source_master, FALSE);
+ source_output_data.destination_source = u->source;
+ /* FIXME
+ source_output_data.flags = PA_SOURCE_OUTPUT_DONT_INHIBIT_AUTO_SUSPEND; */
+
+ pa_proplist_sets(source_output_data.proplist, PA_PROP_MEDIA_NAME, "Echo-Cancel Source Stream");
+ pa_proplist_sets(source_output_data.proplist, PA_PROP_MEDIA_ROLE, "filter");
+ pa_source_output_new_data_set_sample_spec(&source_output_data, &source_ss);
+ pa_source_output_new_data_set_channel_map(&source_output_data, &source_map);
+
+ pa_source_output_new(&u->source_output, m->core, &source_output_data);
+ pa_source_output_new_data_done(&source_output_data);
+
+ if (!u->source_output)
+ goto fail;
+
+ u->source_output->parent.process_msg = source_output_process_msg_cb;
+ u->source_output->push = source_output_push_cb;
+ u->source_output->process_rewind = source_output_process_rewind_cb;
+ u->source_output->update_max_rewind = source_output_update_max_rewind_cb;
+ u->source_output->update_source_requested_latency = source_output_update_source_requested_latency_cb;
+ u->source_output->update_source_latency_range = source_output_update_source_latency_range_cb;
+ u->source_output->update_source_fixed_latency = source_output_update_source_fixed_latency_cb;
+ u->source_output->kill = source_output_kill_cb;
+ u->source_output->attach = source_output_attach_cb;
+ u->source_output->detach = source_output_detach_cb;
+ u->source_output->state_change = source_output_state_change_cb;
+ u->source_output->may_move_to = source_output_may_move_to_cb;
+ u->source_output->moving = source_output_moving_cb;
+ u->source_output->userdata = u;
+
+ u->source->output_from_master = u->source_output;
+
+ /* Create sink input */
+ pa_sink_input_new_data_init(&sink_input_data);
+ sink_input_data.driver = __FILE__;
+ sink_input_data.module = m;
+ pa_sink_input_new_data_set_sink(&sink_input_data, sink_master, FALSE);
+ sink_input_data.origin_sink = u->sink;
+ pa_proplist_sets(sink_input_data.proplist, PA_PROP_MEDIA_NAME, "Echo-Cancel Sink Stream");
+ pa_proplist_sets(sink_input_data.proplist, PA_PROP_MEDIA_ROLE, "filter");
+ pa_sink_input_new_data_set_sample_spec(&sink_input_data, &sink_ss);
+ pa_sink_input_new_data_set_channel_map(&sink_input_data, &sink_map);
+ sink_input_data.flags = PA_SINK_INPUT_VARIABLE_RATE;
+
+ pa_sink_input_new(&u->sink_input, m->core, &sink_input_data);
+ pa_sink_input_new_data_done(&sink_input_data);
+
+ if (!u->sink_input)
+ goto fail;
+
+ u->sink_input->parent.process_msg = sink_input_process_msg_cb;
+ u->sink_input->pop = sink_input_pop_cb;
+ u->sink_input->process_rewind = sink_input_process_rewind_cb;
+ u->sink_input->update_max_rewind = sink_input_update_max_rewind_cb;
+ u->sink_input->update_max_request = sink_input_update_max_request_cb;
+ u->sink_input->update_sink_requested_latency = sink_input_update_sink_requested_latency_cb;
+ u->sink_input->update_sink_latency_range = sink_input_update_sink_latency_range_cb;
+ u->sink_input->update_sink_fixed_latency = sink_input_update_sink_fixed_latency_cb;
+ u->sink_input->kill = sink_input_kill_cb;
+ u->sink_input->attach = sink_input_attach_cb;
+ u->sink_input->detach = sink_input_detach_cb;
+ u->sink_input->state_change = sink_input_state_change_cb;
+ u->sink_input->may_move_to = sink_input_may_move_to_cb;
+ u->sink_input->moving = sink_input_moving_cb;
+ u->sink_input->volume_changed = sink_input_volume_changed_cb;
+ u->sink_input->mute_changed = sink_input_mute_changed_cb;
+ u->sink_input->userdata = u;
+
+ u->sink->input_to_master = u->sink_input;
+
+ pa_sink_input_get_silence(u->sink_input, &silence);
+
+ u->source_memblockq = pa_memblockq_new(0, MEMBLOCKQ_MAXLENGTH, 0,
+ pa_frame_size(&source_ss), 1, 1, 0, &silence);
+ u->sink_memblockq = pa_memblockq_new(0, MEMBLOCKQ_MAXLENGTH, 0,
+ pa_frame_size(&sink_ss), 1, 1, 0, &silence);
+
+ pa_memblock_unref(silence.memblock);
+
+ if (!u->source_memblockq || !u->sink_memblockq) {
+ pa_log("Failed to create memblockq.");
+ goto fail;
+ }
+
+ /* our source and sink are not suspended when we create them */
+ u->active_mask = 3;
+
+ if (u->adjust_time > 0)
+ u->time_event = pa_core_rttime_new(m->core, pa_rtclock_now() + u->adjust_time, time_callback, u);
+
+ if (u->save_aec) {
+ pa_log("Creating AEC files in /tmp");
+ u->captured_file = fopen("/tmp/aec_rec.sw", "wb");
+ if (u->captured_file == NULL)
+ perror ("fopen failed");
+ u->played_file = fopen("/tmp/aec_play.sw", "wb");
+ if (u->played_file == NULL)
+ perror ("fopen failed");
+ u->canceled_file = fopen("/tmp/aec_out.sw", "wb");
+ if (u->canceled_file == NULL)
+ perror ("fopen failed");
+ }
+
+ pa_sink_put(u->sink);
+ pa_source_put(u->source);
+
+ pa_sink_input_put(u->sink_input);
+ pa_source_output_put(u->source_output);
+
+ pa_modargs_free(ma);
+
+ return 0;
+
+fail:
+ if (ma)
+ pa_modargs_free(ma);
+
+ pa__done(m);
+
+ return -1;
+}
+
+int pa__get_n_used(pa_module *m) {
+ struct userdata *u;
+
+ pa_assert(m);
+ pa_assert_se(u = m->userdata);
+
+ return pa_sink_linked_by(u->sink) + pa_source_linked_by(u->source);
+}
+
+void pa__done(pa_module*m) {
+ struct userdata *u;
+
+ pa_assert(m);
+
+ if (!(u = m->userdata))
+ return;
+
+ /* See comments in source_output_kill_cb() above regarding
+ * destruction order! */
+
+ if (u->time_event)
+ u->core->mainloop->time_free(u->time_event);
+
+ if (u->source_output)
+ pa_source_output_unlink(u->source_output);
+ if (u->sink_input)
+ pa_sink_input_unlink(u->sink_input);
+
+ if (u->source)
+ pa_source_unlink(u->source);
+ if (u->sink)
+ pa_sink_unlink(u->sink);
+
+ if (u->source_output)
+ pa_source_output_unref(u->source_output);
+ if (u->sink_input)
+ pa_sink_input_unref(u->sink_input);
+
+ if (u->source)
+ pa_source_unref(u->source);
+ if (u->sink)
+ pa_sink_unref(u->sink);
+
+ if (u->source_memblockq)
+ pa_memblockq_free(u->source_memblockq);
+ if (u->sink_memblockq)
+ pa_memblockq_free(u->sink_memblockq);
+
+ if (u->ec->pp_state)
+ speex_preprocess_state_destroy(u->ec->pp_state);
+
+ if (u->ec) {
+ if (u->ec->done)
+ u->ec->done(u->ec);
+
+ pa_xfree(u->ec);
+ }
+
+ if (u->asyncmsgq)
+ pa_asyncmsgq_unref(u->asyncmsgq);
+
+ pa_xfree(u);
+}
diff --git a/src/modules/echo-cancel/speex.c b/src/modules/echo-cancel/speex.c
new file mode 100644
index 00000000..72c52680
--- /dev/null
+++ b/src/modules/echo-cancel/speex.c
@@ -0,0 +1,115 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2010 Wim Taymans <wim.taymans@gmail.com>
+
+ Contributor: Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulsecore/modargs.h>
+#include "echo-cancel.h"
+
+/* should be between 10-20 ms */
+#define DEFAULT_FRAME_SIZE_MS 20
+/* should be between 100-500 ms */
+#define DEFAULT_FILTER_SIZE_MS 200
+
+static const char* const valid_modargs[] = {
+ "frame_size_ms",
+ "filter_size_ms",
+ NULL
+};
+
+static void pa_speex_ec_fixate_spec(pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map)
+{
+ source_ss->format = PA_SAMPLE_S16NE;
+
+ *sink_ss = *source_ss;
+ *sink_map = *source_map;
+}
+
+pa_bool_t pa_speex_ec_init(pa_core *c, pa_echo_canceller *ec,
+ pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map,
+ uint32_t *blocksize, const char *args)
+{
+ int framelen, y, rate;
+ uint32_t frame_size_ms, filter_size_ms;
+ pa_modargs *ma;
+
+ if (!(ma = pa_modargs_new(args, valid_modargs))) {
+ pa_log("Failed to parse submodule arguments.");
+ goto fail;
+ }
+
+ filter_size_ms = DEFAULT_FILTER_SIZE_MS;
+ if (pa_modargs_get_value_u32(ma, "filter_size_ms", &filter_size_ms) < 0 || filter_size_ms < 1 || filter_size_ms > 2000) {
+ pa_log("Invalid filter_size_ms specification");
+ goto fail;
+ }
+
+ frame_size_ms = DEFAULT_FRAME_SIZE_MS;
+ if (pa_modargs_get_value_u32(ma, "frame_size_ms", &frame_size_ms) < 0 || frame_size_ms < 1 || frame_size_ms > 200) {
+ pa_log("Invalid frame_size_ms specification");
+ goto fail;
+ }
+
+ pa_speex_ec_fixate_spec(source_ss, source_map, sink_ss, sink_map);
+
+ rate = source_ss->rate;
+ framelen = (rate * frame_size_ms) / 1000;
+ /* framelen should be a power of 2, round down to nearest power of two */
+ y = 1 << ((8 * sizeof (int)) - 2);
+ while (y > framelen)
+ y >>= 1;
+ framelen = y;
+
+ *blocksize = framelen * pa_frame_size (source_ss);
+
+ pa_log_debug ("Using framelen %d, blocksize %u, channels %d, rate %d", framelen, *blocksize, source_ss->channels, source_ss->rate);
+
+ ec->params.priv.speex.state = speex_echo_state_init_mc (framelen, (rate * filter_size_ms) / 1000, source_ss->channels, source_ss->channels);
+
+ if (!ec->params.priv.speex.state)
+ goto fail;
+
+ speex_echo_ctl(ec->params.priv.speex.state, SPEEX_ECHO_SET_SAMPLING_RATE, &rate);
+
+ pa_modargs_free(ma);
+ return TRUE;
+
+fail:
+ if (ma)
+ pa_modargs_free(ma);
+ return FALSE;
+}
+
+void pa_speex_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
+ speex_echo_cancellation(ec->params.priv.speex.state, (const spx_int16_t *) rec, (const spx_int16_t *) play, (spx_int16_t *) out);
+}
+
+void pa_speex_ec_done(pa_echo_canceller *ec) {
+ if (ec->params.priv.speex.state)
+ speex_echo_state_destroy(ec->params.priv.speex.state);
+ ec->params.priv.speex.state = NULL;
+}
generated by cgit (git 2.34.1) at 2026-01-17 11:34:20 +0000